Mobility Beyond the Enterprise
With Cisco’s mobile collaboration solutions, mobility users can handle calls to their enterprise directory number, not only on their desk phone, but also on one or more remote phones. Mobility users can also make calls from a remote phone as if they are dialing inside the enterprise. In addition, mobility users can take advantage of enterprise features such as hold, transfer, and conference as well as enterprise applications such as voicemail, conferencing, and presence on their mobile phones. This ensures continued productivity for users even when they are traveling outside the organization.
Further, with dual-mode phones that provide connectivity to the mobile voice and data provider network as well as the 802.11 WLAN, users not only have the ability to leverage enterprise applications while away from the enterprise, but they can also leverage the enterprise telephony infrastructure when inside the enterprise or remotely attached to the enterprise network to make and receive calls without incurring mobile voice network per-minute charges.
The fixed mobile convergence (FMC) mobility functionality delivered within the Cisco Unified Mobility solution is provided through Cisco Unified CM and can be used in conjunction with Cisco mobile clients and devices such as Cisco Jabber.
Cisco Unified Mobility provides the following mobility application functionality:
- Single Number Reach (SNR)
Single Number Reach provides users with the ability to be reached at a single enterprise phone number that rings on both their IP desk phone and their mobile phone simultaneously. SNR users can pick up an incoming call on either their desk or mobile phones and at any point can move the in-progress call from one of these phones to the other without interruption.
Mid-call features allow a user to invoke hold, resume, transfer, conferencing, and directed call park features from their mobile phone during in-progress mobility calls. These features are invoked from the mobile phone keypad and take advantage of enterprise media resources such as music on hold and conference bridges.
- Single Enterprise Voicemail Box
Single Enterprise Voicemail box provides mobile voicemail avoidance capabilities and ensures that any unanswered calls made to the user's enterprise number and extended to the user's mobile phone will end up in the enterprise voicemail system rather than in a mobile voicemail system. This provides a single consolidated voicemail box for all business calls and eliminates the need for users to check multiple voicemail systems for messages.
- Mobile Voice Access and Enterprise Feature Access two-stage dialing
Mobile Voice Access and Enterprise Feature Access two-stage dialing provide mobile users with the ability to make calls from their mobile phone as if they were calling from their enterprise IP desk phone. These features provide a cost savings in terms of toll charges for long distance or international calls as well as calls to internal non-DID extensions on the system that would not normally be reachable from outside the enterprise. These two-stage dialing features also provide the enterprise with an easy way to track phone calls made by users via a uniform and centrally located set of call detail records. Furthermore, these features provide the ability to mask a user's mobile phone number when sending outbound caller ID. Instead, the user's enterprise number is sent as caller ID. This ensures that returned calls to the user are made to the enterprise number, thus resulting in enterprise call anchoring.
Cisco mobile clients and devices provide the ability to attach to both the mobile provider network and 802.11 wireless networks for voice and data connectivity. This enables users to leverage both enterprise call control and in some cases mobile network call control from a single device. By leveraging the enterprise telephony infrastructure for making and receiving calls whenever possible and, in the case of dual-mode phones, falling back to the mobile voice network only when enterprise connectivity is unavailable, mobile clients and devices can help reduce telephony costs. Dual-mode phones and the clients that run on them also provide a handoff mechanism so that in-progress voice calls can be moved easily between the WLAN and mobile voice interfaces as a user moves out of the enterprise.
In addition to enabling mobile devices to make voice or video calls over IP via 802.11 WLAN or mobile data networks, Cisco mobile clients enable automated enterprise dialing using the Dial via Office feature. Dial via Office calls are set up using SIP signaling over the IP network, while the media path is over the mobile voice network and the PSTN. Cisco mobile clients and devices also provide other unified communications services such as corporate directory access, presence and instant messaging (IM). These devices and clients enable mobile users to remain productive whether inside or outside the enterprise by providing access to collaboration applications while at the same time enabling users to make and receive enterprise calls from their mobile devices, whether outside the enterprise over public or private WiFi hot spots or the mobile data network, or inside the enterprise and over the WLAN network.
This section begins with a discussion of Unified Mobility features, functionality, and design and deployment considerations. Given the various benefits of Unified Mobility and the fact that mobile clients and devices can be integrated to take advantage of the features provided, this discussion paves the way for examination of mobile client applications such as Cisco Jabber. This section also includes a discussion of architecture, functionality, and design and deployment implications for the following mobility applications and features:
Cisco Unified Mobility
Cisco Unified Mobility refers to the native mobility functionality within the Cisco Unified CM and includes the Single Number Reach, Mobile Voice Access, and Enterprise Feature Access features.
Unified Mobility functionality depends on the appropriate configuration of Unified CM. For this reason, it is important to understand the nature of this configuration as well as the logical components.
Figure 21-17 illustrates the configuration requirements for Unified Mobility. First, as for all users, a mobility user’s enterprise phone is configured with appropriate line-level settings such as directory number, partition, and calling search space. In addition, the device-level settings of the enterprise phone include parameters such as device pool, common device configuration, calling search space, media resource group list, and user and network hold audio sources. All of these line and device settings on the user’s enterprise phone affect the call routing and music on hold (MoH) behavior for incoming and outgoing calls.
Next, a remote destination profile must be configured for each mobility user in order for them to take advantage of Unified Mobility features. The remote destination profile is configured at the line level with the same directory number, partition, and calling search space as the user’s enterprise phone line. This results in a shared line between the remote destination profile and the enterprise phone. The remote destination profile configuration includes device pool, calling search space, rerouting calling search space, and user and network hold audio source parameters. The remote destination profile should be thought of as a virtual phone whose configuration mirrors the user’s line-level enterprise phone settings, but whose profile-level configuration combined with the line-level settings determines the call routing and MoH behavior that the user’s remote destination phone will inherit. The user’s enterprise directory number, which is shared between the remote destination profile and the enterprise phone, allows calls to that number to be extended to the user’s remote destination.
Figure 21-17 Cisco Unified Mobility Configuration Architecture
As further shown in Figure 21-17, a mobility user can have one or more remote destinations configured and associated with their remote destination profile. A remote destination represents a single PSTN phone number where a user can be reached. A user can have up to 10 remote destinations defined. Call routing timers can be configured for each remote destination to adjust the amount of time a call will be extended to a particular remote phone, as well as the amount of time to wait before extending the call and the amount of time that must pass before a call can be answered at the remote phone. Mobility users can also configure filters for each remote destination to allow or deny calls from certain phone numbers to be extended to that remote phone.
Single Number Reach
The Single Number Reach (SNR) feature allows an incoming call to an enterprise user to be offered to the user's IP desk phone as well as up to 10 configurable remote destinations. Typically a user's remote destination is their mobile or cellular telephone. Once the call is offered to both the desktop and remote destination phone(s), the user can answer at any of those phones. Upon answering the call on one of the remote destination phones or on the IP desk phone, the user has the option to hand off or pick up the call on the other phone.
Single Number Reach Functionality
Figure 21-18 illustrates a basic Single Number Reach call flow. In this example, Phone A on the PSTN calls an SNR user's enterprise directory number (DN) 408-555-1234 (step 1). The call comes into the enterprise PSTN gateway and is extended through Unified CM to the IP phone with DN 408-555-1234 (step 2), and this phone begins to ring. The call is also extended to the user’s Remote Destination Profile, which shares the same DN (step 3). In turn, a call is placed to the remote destination associated with the user's remote destination profile (in this case 408-555-7890) (step 4). The outgoing call to the remote destination is routed through the PSTN gateway (step 5). Finally the call rings at the remote destination PSTN phone with number 408 555-7890 (step 6). The call can then be answered at either phone.
Figure 21-18 Single Number Reach
Typically a Single Number Reach user's configured remote destination is their mobile phone on a mobile voice or cellular provider network; however, any destination reachable by means of the PSTN can be configured as a user's remote destination. Furthermore, an SNR user can have up to 10 remote destinations configured, so an incoming call could potentially ring as many as 10 PSTN phones as well as the user's desk phone. Once the call is answered at the desk phone or at a remote destination phone, any other call legs that have been extended to ring additional remote destinations or the desk phone (if not answered at the desk phone) will be cleared. If the incoming call is answered at the remote destination, the voice media path will be hairpinned within the enterprise PSTN gateway utilizing two gateway ports. This utilization must be considered when deploying the SNR feature.
Note In order for Single Number Reach to work as in Figure 21-18, ensure that the user-level Enable Mobility check box under the End User configuration page has been checked and that at least one of the user's configured remote destinations has the Enable Single Number Reach check box checked.
Desk Phone Pickup
As illustrated in Figure 21-19, once a user answers a Single Number Reach call at the remote destination device (step 1: in this case, 408 555-7890), at any point the user can hang up the call at the remote destination and pick it up again at their desk phone by simply pressing the Resume softkey on the desk phone (step 2: at DN 408 555-1234 in this case). The call resumes between the original caller at Phone A and the desk phone (step 3).
Figure 21-19 Desk Phone Pickup
Desk phone pickup can be performed whenever an enterprise-anchored call is in progress at a configured remote destination phone and that phone hangs up the call.
Note An enterprise-anchored call refers to any call that has at least one call leg connected through an enterprise PSTN gateway and that originated either from a remote destination to an enterprise DID or from Single Number Reach, Mobile Voice Access, Enterprise Feature Access, or Intelligent Session Control.
The option to pick up or resume the call at the desk phone is available for a certain amount of time. For this reason, it is good practice for the Single Number Reach user to ensure that the calling phone hangs up before the remote destination phone is hung up. This ensures that the call cannot be resumed at the desk phone by someone else. By default, the call remains available for pickup at the desk phone for 10 seconds after the remote destination phone hangs up; however, this time is configurable and can be set from 0 to 30000 milliseconds on a per-user basis by changing the Maximum Wait Time for Desk Pickup parameter under the End User configuration page. Desk phone pickup can also be performed after invoking the mid-call hold feature at the remote destination phone. However, in these cases, the Maximum Wait Time for Desk Pickup parameter setting has no effect on the amount of time the call will be available for pickup. A call placed on mid-call hold will remain on hold and be available for desk phone pickup until manually resumed at either the remote or desktop phone.
Another method for performing desk phone pickup is to use the mid-call session handoff feature. This mid-call feature is invoked by manually keying *74, the default enterprise feature access code for session handoff, which in turn generates a DTMF sequence back to Unified CM. When this feature is invoked, Unified CM sends a new call to the user's enterprise desk phone. Once this new call is flashing or ringing at the desk phone, the user then must answer the call to complete the session handoff.
The benefit of this desk phone pickup method over other methods (such as hanging up the call at the mobile phone or using the mid-call hold feature) is that the conversation between the user and the far-end phone is maintained throughout the handoff process. Once the *74 sequence has been keyed, the user can continue the conversation because the handoff call is sent to the user's desk phone. When the user answers the call at the desk phone, the call legs are shuffled so that the call leg to the far-end is connected to the new call leg created at the desk phone, thus resulting in an uninterrupted or near-instantaneous cut-through of the audio path. The original call leg at the mobile device is subsequently cleared.
Unlike the hang-up method for invoking desk phone pickup, where the end-user’s Maximum Wait Time for Desk Pickup setting determines how long the call will be available for pickup at the desk phone, with session handoff the Session Handoff Alerting Timer service parameter determines the amount of time the call will ring or flash at the desk phone before the handoff call is cleared. The default handoff alerting time is 10 seconds. Further, with session handoff, any call forward settings configured on the desk phone do not get invoked. As a result, the handoff feature does not forward to voicemail or any other call-forward destination. If a call is not answered by the end of Session Handoff Alerting Timer period, then the call is cleared and the Remote In Use state is removed from the user's desk phone line. However, in this scenario the original call is maintained at the mobile phone.
For additional information about session handoff and other mid-call features, see Mid-Call Features.
Remote Destination Phone Pickup
Figure 21-20 illustrates Single Number Reach remote destination phone pickup functionality. Assuming Phone A calls the SNR user's enterprise DN 408 555-1234 and the call is answered at the user's desk phone and is in progress (step 1), the user must push the Mobility softkey. Assuming the SNR feature is enabled for this phone and remote destination pickup is available, the user presses the Select softkey (step 2). A call is generated to the user's remote destination phone (in this case, 408 555-7890), and the remote phone begins to ring. Once the call is answered at the remote phone, the call resumes between Phone A and the SNR user's remote phone with number 408 555-7890 (step 3).
Figure 21-20 Remote Destination Phone Pickup
When a Single Number Reach user has multiple remote destinations configured, each remote destination will ring when the Select softkey is pressed, and the user can answer the desired phone.
Note In order for remote destination phone pickup to work as in Figure 21-20, ensure that at least one of the user's configured remote destinations has the Mobile Phone check box checked. In addition, the Mobility softkey must be configured for all mobility users by adding the softkey to each user's associated desk phone softkey template. Failure to check the Mobile Phone check box and to make the Mobility softkey available to mobility users will prevent the use of remote destination phone pickup functionality.
Note Cisco TelePresence System C, EX, MX, SX, and TX Series video endpoints do not support remote destination pickup as described above. These endpoints do not expose a mobility softkey or the "Send call to Mobile Phone" option to the user. Therefore, these endpoints are unable to send in-progress calls to the mobile device using remote destination pickup.
As illustrated in Figure 21-21, once a user answers a Single Number Reach call at the remote destination device (step 1: in this case, 408 555-7890), the user can invoke mid-call features such as hold, resume, transfer, conference, directed call park, and session handoff by sending DTMF digits from the remote destination phone to Unified CM via the enterprise PSTN gateway (step 2). When the mid-call feature hold, transfer, conference, or directed call park is invoked, MoH is forwarded from Unified CM to the held party (step 3: in this case, Phone A). In-progress calls can be transferred to another phone or directed call park number, or additional phones can be conferenced using enterprise conference resources (step 4).
Figure 21-21 Mobility Mid-Call Feature
Mid-call features are invoked at the remote destination phone by a series of DTMF digits forwarded to Unified CM. Once received by Unified CM, these digit sequences are matched to the configured Enterprise Feature Access Codes for Hold, Exclusive Hold, Resume, Transfer, Conference, and Session Handoff, and the appropriate function is performed.
Note To enable the Directed Call Park mid-call feature, you must configure Cisco Unified CM with directed call park numbers and call park retrieval prefixes.
Note In order to perform the transfer, conference, and directed call park mid-call features, a second call leg is generated by the remote destination phone to a system-configured Enterprise Feature Access DID that answers the call, takes user input (including PIN number, mid-call feature access code, and target number), and then creates the required call leg to complete the transfer, conference, or directed call park operation.
With the mid-call session handoff feature, MoH is not forwarded to the far-end because the far-end is never placed on hold. Instead, the original audio path is maintained until the mobile user answers the handoff call at the desk phone. Once the call is answered, the call legs are shuffled at the enterprise gateway and the audio path is maintained.
Mid-call features are invoked by manually keying the feature access codes and entering the appropriate key sequences. Table 21-2 indicates the required key sequences for invoking mid-call features.
Table 21-2 Manual Mid-Call Feature Key Sequences
Enterprise Feature Access Code (default)
1. Enter: *82 (Exclusive Hold)
2. Make new call to Enterprise Feature Access DID.
3. On connect, enter:
<PIN_number> # *84 # <Transfer_Target/DN> #
4. Upon answer by transfer target (for consultive transfer) or upon ringback (for early attended transfer), enter: *84
Directed Call Park
1. Enter: *82 (Exclusive Hold)
2. Make new call to Enterprise Feature Access DID.
3. On connect, enter:
<PIN_number> # *84 # <Directed_Call_Park_Number> # *84 #
Note To retrieve a parked call, the user must use Mobile Voice Access or Enterprise Feature Access Two-Stage Dialing to place a call to the directed call park number. When entering the directed call park number to be dialed, it must be prefixed with the appropriate call park retrieval prefix.
1. Enter: *82 (Exclusive Hold)
2. Make new call to Enterprise Feature Access DID.
3. On connect enter:
<PIN_number> # *85 # <Conference_Target/DN> #
4. Upon answer by conference target, enter: *85
1. Enter: *74
2. Answer at the desk phone upon ring and/or flash.
Note Media resource allocation for mid-call features such as hold and conference is determined by the Remote Destination Profile configuration or, in the case of dual-mode phones and Unified Mobile Communicator, the device configuration. The media resource group list (MRGL) of the device pool configured for the Remote Destination Profile or the mobile client device is used to allocate a conference bridge for the conferencing mid-call feature. The User Hold Audio Source and Network Hold MoH Audio Source settings of the Remote Destination Profile or the mobile client device, in combination with the media resource group list (MRGL) of the device pool, is used to determine the appropriate MoH stream to be sent to a held device.
Mobile Voicemail Avoidance with Single Enterprise Voicemail Box
An additional consideration with Cisco Unified Mobility Single Number Reach is mobile voicemail avoidance. The single enterprise voicemail box feature ensures that all unanswered enterprise business calls end up at the enterprise voicemail system. This prevents a user from having to check multiple mailboxes (enterprise, mobile, home, and so forth) for calls to their enterprise phone number that are unanswered. This feature provides two methods for avoiding mobile or non-enterprise voicemail:
- Timer Control method — With this method the system relies on a set of timers (one per remote destination) in conjunction with system call-forward timers to ensure that, when and if a call is forwarded to a voicemail system on ring-no-answer, the enterprise voicemail system receives the call.
- User Control method — With this method the system relies on a DTMF confirmation tone from the remote destination when the call is answered to determine if the call was received by the user or a non-enterprise voicemail system.
System settings determine whether the timer control or user control method is used. The method used can be set globally via the Voicemail Selection Policy service parameter or for individual remote destinations via the Single Number Reach Voicemail Policy. By default the system and all remote destinations use the timer control method
Timer Control Mobile Voicemail Avoidance
For this method, the system relies on a set of timers on the Remote Destination configuration page. The purpose of these timers is to ensure that, when and if a call is forwarded to a voicemail system on ring-no-answer, the call is forwarded to the enterprise voicemail system rather than any remote destination voicemail system. These timers in conjunction with other system forward-no-answer timers should be configured to avoid non-enterprise voicemail systems as follows:
- Ensure the system forward-no-answer time is shorter at the desk phone than at the remote destination phones.
To do so, ensure that the global Forward No Answer Timer field in Unified CM or the No Answer Ring Duration field under the individual phone line is configured with a value that is less than the amount of time a remote destination phone will ring before forwarding to the mobile voicemail system. In addition, the Delay Before Ringing Timer parameter under the Remote Destination configuration page can be used to delay the ringing of the remote destination phone in order to further lengthen the amount of time that must pass before a remote destination phone will forward to its own mobile voicemail box. However, when adjusting the Delay Before Ringing Timer parameter, take care to ensure that the global Unified CM Forward No Answer Timer (or the line-level No Answer Ringer Duration field) is set sufficiently high enough so that the mobility user has time to answer the call on the remote destination phone. The Delay Before Ringing Timer parameter can be set for each remote destination and is set to 4,000 milliseconds by default.
- Ensure that the remote destination device stops ringing before the incoming call is forwarded to the mobile voicemail system.
You can accomplish this with the Answer Too Soon and Answer Too Late timers for each remote destination. First the Answer Too Soon Timer parameter under the Remote Destination configuration page should be configured with a value that is more than the amount of time it takes a call extended to a powered-off or out-of-range mobile phone to be forwarded to the mobile voicemail system. By default this timer is set 1,500 milliseconds (or 1.5 seconds). If the call is answered before the Answer Too Soon Timer expires, the system will disconnect the call leg to the remote destination. This ensures that calls forwarded immediately to the mobile voicemail system will not be connected, but those answered by the user after ring-in are connected.
Next configure the Answer Too Late Timer parameter under the Remote Destination configuration page with a value that is less than the amount of time that a remote destination phone will ring before forwarding to its voicemail box. By default this timer is set to 19,00 milliseconds (or 19 seconds). If the call is not answered before this timer expires, the system will disconnect the call leg to the remote destination. This ensures that the remote destination phone stops ringing before the call is forwarded to the mobile voicemail system.
Note Incoming calls to a remote destination that are manually diverted by the mobility user can end up in the mobile voicemail box if the manual diversion occurs after the Answer Too Soon timer has expired. To prevent this from happening, mobility users should be configured for the user control method or advised to ignore or silence the ringing of incoming calls they wish to divert to voicemail. This will ensure that unanswered calls always end up in the enterprise voicemail system.
Note In most deployment scenarios, the default Delay Before Ringing Timer, Answer Too Late Timer, and Answer Too Soon Timer values are sufficient and do not need to be changed.
User Control Mobile Voicemail Avoidance
For this method, the system relies on DTMF confirmation tone from the remote destination when the call is answered. If a DTMF tone is received by the system, then the system knows that the user answered the call and pressed a key to generate the DTMF tone. On the other hand, if the DTMF tone is not received by the system, the system assumes the call leg was answered by a non-enterprise voicemail system and it disconnects the call leg.
When the user control method is enabled, on answer the end user will hear an audio prompt requesting that they press a key pad button to generate a DTMF tone. By default the audio prompt is played to the user one second after the call is answered. The user may not hear the audio prompt if they press the keypad to generate a DTMF tone immediately upon answering. The audio prompt is played only on the remote destination call leg and therefore the far-end party will not hear this prompt. Once the audio prompt is played to the user, by default the system will wait 5 seconds to receive the DTMF tone. If the tone is not received, the system disconnects the call leg but continues to ring the user's other configured devices until the call is answered by the user or forwarded to the enterprise voicemail system.
Note The user control mobile voicemail avoidance method is completely dependent on successful relay of the DTMF tone from the remote destination on the mobile voice network or PSTN all the way to Unified CM. The DTMF tone must be sent out-of-band to Unified CM. If DTMF relay is not properly configured on the network and system, DTMF will not be received and all call legs to remote destinations relying on the user control method will be disconnected. The system administrator should ensure proper DTMF interoperation and relay across the enterprise telephony network prior to enabling the user control method. If DTMF cannot be effectively relayed from the PSTN to Unified CM, then the timer control mobile voicemail avoidance method should be used instead.
Enabling and Disabling Single Number Reach
The Single Number Reach (SNR) feature can be enabled or disabled by using one of the following methods:
- Cisco Unified CM Administration or Cisco Unified CM Self Care Portal for end users
An administrator or user unchecks the Enable Single Number Reach box to disable, or checks the Enable Single Number Reach box to enable, the feature. This is done per remote destination.
- Mobile Voice Access or Enterprise Feature Access
A Mobility-enabled user dials into the Mobile Voice Access or Enterprise Feature Access DID and, after entering appropriate credentials, enters the digit 2 to enable or 3 to disable. With Mobile Voice Access, the user is prompted to enable or disable SNR for a single remote destination or all of their remote destinations. With Enterprise Feature Access, the user can enable or disable SNR only for the remote destination device from which they are calling.
- Desk phone Mobility softkey or icon
The user presses the Mobility softkey when the phone is in the on-hook state and selects either Enable Mobile Connect or Disable Mobile Connect. On some phone models the user touches the mobility icon and then selects Off to disable Single Number Reach. Alternatively, the user can select Ring only this phone. To enable Single Number Reach again the user selects Ring all devices. With any of these methods, Single Number Reach is enabled or disabled for all of the user's remote destinations.
Note The dialog box that appears when the Mobility softkey is pressed as described above uses the old feature name, Mobile Connect, rather than the new feature name, Single Number Reach. The feature and enable/disable functionality are the same.
Access Lists for Allowing or Blocking Single Number Reach Calls
Access lists can be configured within Cisco Unified CM and associated to a remote destination. Access lists are used to allow or block inbound calls (based on incoming caller ID) from being extended to a mobility-enabled user's remote destinations. Furthermore, these access lists are invoked based on the time of day.
Access lists are configured for mobility-enabled users as either blocked or allowed. Access lists contain one or more members or filters consisting of a specific number or number mask, and the filters are compared against the incoming caller ID of the calling party. In addition to containing specific number strings or number masks for matching caller ID, access lists can also contain a filter for incoming calls where the caller ID is not available or is set to private. A blocked access list contains an implicit "allow all" at the end of the list so that calls from any numbers entered in the access list will be blocked but calls from all other numbers will be allowed. An allowed access list contains an implicit "deny all" at the end of the list so that calls from any numbers entered in the access list will be allowed but calls from all other numbers will be blocked.
Once configured access lists are associated with a configured Ring Schedule under the Remote Destination configuration screen, the configured Ring Schedule in combination with the selected access list provides time-of-day call filtering for Single Number Reach calls on a per-remote-destination basis. Access lists and Ring Schedules can be configured and associated to a remote destination by an administrator using the Cisco Unified CM Administration interface or by an end user using the Cisco Unified CM Self Care Portal.
Single Number Reach Architecture
The architecture of the Single Number Reach (SNR) feature is as important to understand as its functionality. Figure 21-22 depicts the message flows and architecture required for SNR. The following sequence of interactions and events can occur between Unified CM, the SNR user, and the SNR user's desk phone:
1. The SNR phone user who wishes to either enable or disable the SNR feature or to pick up an in-progress call on their remote destination phone pushes the Mobility softkey on their desk phone (see step 1 in Figure 21-22).
2. Unified CM returns the SNR status (On or Off) and offers the user the ability to select the Send Call to Mobile Phone option when the phone is in the Connected state, or it offers the user the ability to enable or disable the Mobile Connect status when the phone is in the On Hook state (see step 2 in Figure 21-22).
3. Single Number Reach users can use the Unified CM Self Care Portal to configure their own mobility settings via the web-based configuration pages at
https:// <Unified-CM_Server_IP_Address> /ucmuser/
where <Unified-CM_Server_IP_Address> is the IP address of the Unified CM publisher server (see step 3 in Figure 21-22).
Figure 21-22 Single Number Reach Architecture
High Availability for Single Number Reach
The Single Number Reach feature relies on the following components:
- Unified CM servers
- PSTN gateway
Each component must be redundant or resilient in order for Single Number Reach to continue functioning fully during various failure scenarios.
Unified CM Server Redundancy
The Unified CM server is required for the Single Number Reach feature. Unified CM server failures are non-disruptive to SNR functionality, assuming phone and gateway registrations are made redundant using Unified CM Groups.
In order for SNR users to use the Unified CM Self Care Portal web interface to configure their mobility settings (remote destinations and access lists), the Unified CM publisher server must be available. If the publisher is down, users will not be able to change mobility settings. Likewise, administrators will be unable to make mobility configuration changes to Unified CM; however, existing mobility configurations and functionality will continue. Finally, changes to SNR status must be written by the system on the Unified CM publisher server; if the Unified CM publisher is unavailable, then enabling or disabling SNR will not be possible.
PSTN Gateway Redundancy
Because the Single Number Reach feature relies on the ability to extend additional call legs to the PSTN to reach the SNR users’ remote destination phones, PSTN gateway redundancy is important. Should a PSTN gateway fail or be out of capacity, the SNR call cannot complete. Typically, enterprise IP telephony dial plans provide redundancy for PSTN access by providing physical gateway redundancy and call re-routing capabilities as well as enough capacity to handle expected call activity. Assuming that Unified CM has been configured with sufficient capacity, multiple gateways, and route group and route list constructs for call routing resiliency, the SNR feature can rely on this redundancy for uninterrupted functionality.
Mobile Voice Access and Enterprise Feature Access
Mobile Voice Access (also referred to as System Remote Access) and Enterprise Feature Access two-stage dialing are features built on top of the Single Number Reach application. Both features allow a mobility-enabled user who is outside the enterprise to make a call as though they are directly connected to Unified CM. This functionality is commonly referred to as Direct Inward System Access (DISA) in traditional telephony environments. These features benefit the enterprise by limiting toll charges and consolidating phone billing directly to the enterprise rather than billing to each mobile user. In addition, these features allow the users to mask their mobile phone or remote destination numbers when sending outbound caller ID. Instead, the user's enterprise directory number is sent as caller ID. This ensures that returned calls to the user are made to the enterprise number, thus resulting in enterprise call anchoring. These features also enable mobile users to dial internal extensions or non-DID enterprise numbers that would not normally be reachable from outside the enterprise.
Mobile Voice Access is accessed by calling a system-configured DID number that is answered and handled by an H.323 or SIP VoiceXML (VXML) gateway. The VoiceXML gateway plays interactive voice response (IVR) prompts to the Mobile Voice Access user, requesting user authentication and input of a number to be dialed via the user phone keypad.
Enterprise Feature Access functionality includes the previously discussed mid-call transfer and conference features as well as two-stage dialing functionality. Two-stage dialing works the same way as Mobile Voice Access, but without the IVR prompts. The system-configured Enterprise Feature Access DID is answered by Unified CM. The user then uses the phone keypad or Smart Phone softkeys to input authentication and the number to be dialed. These inputs are received without prompts.
With both the Mobile Voice Access and Enterprise Feature Access two-stage dialing features, once the call to the input number is connected, users can invoke mid-call features or pick up the call on their desk phones just as with a Single Number Reach call. This is possible because the call is anchored at the enterprise gateway.
Mobile Voice Access IVR VoiceXML Gateway URL
The Mobile Voice Access feature requires the Unified CM VoiceXML application to reside on the H.323 or SIP gateway. The URL used to load this application is:
http:// <Unified-CM-Publisher_IP-Address> :8080/ccmivr/pages/IVRMainpage.vxml
where <Unified-CM-Publisher_IP-Address> is the IP address of the Unified CM publisher node.
Mobile Voice Access Functionality
Figure 21-23 illustrates a Mobile Voice Access call flow. In this example, the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 408-555-2345 (step 1).
The call comes into the enterprise PSTN H.323 or SIP gateway, which also serves as the VoiceXML gateway (step 2).
Note Native VoiceXML support is not available with Cisco IOS XE; therefore, the Cisco 4000 Series Integrated Services Router (ISR) cannot be deployed as a VoiceXML gateway for Mobile Voice Access. Instead a Cisco IOS gateway supporting native VXML must be used.
The user is prompted via IVR to enter their numeric user ID (followed by the # sign), PIN number (followed by the # sign), and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. In this case, the user enters 9 1 972 555 3456 as the number they wish to reach (followed by the # sign).
Note If the PSTN phone from which the Mobile Voice Access user is calling is configured as a Single Number Reach remote destination for that user and the incoming caller ID can be matched against this remote destination by Unified CM, the user does not have to enter their numeric user ID. Instead they will be prompted to enter just the PIN number.
In the meantime, Unified CM has forwarded IVR prompts to the gateway, the gateway has played these prompts to the user, and the gateway has collected user input including the numeric ID and PIN number of the user. This information is forwarded to Unified CM for authentication and to generate the call to 9 1 972 555 3456 (step 3). After authenticating the user and receiving the number to be dialed, Unified CM generates a call via the user's Remote Destination Profile (step 4). The outbound call to 972 555-3456 is routed via the PSTN gateway (step 5). Finally, the call rings at the PSTN destination phone with number 972 555-3456 (step 6).
Figure 21-23 Mobile Voice Access
Note In order for Mobile Voice Access to work as in Figure 21-23, ensure that the system-wide Enable Mobile Voice Access service parameter is set to True and that the per-user Enable Mobile Voice Access check box on the End User configuration page is also checked.
Note The Mobile Voice Access feature relies on the Cisco Unified Mobile Voice Access Service, which must be activated manually from the Unified CM Serviceability configuration page. This service can be activated on the publisher node only.
Note If the PSTN gateway is a Cisco 4000 Series ISR which does not support native VoiceXML, the VoiceVXML functionality required for Mobile Voice Access must be offloaded to an H.323 Cisco IOS gateway with native VoiceXML support using the hairpinning method of deployment as described in the next section.
Mobile Voice Access Using Hairpinning
In deployments where the enterprise PSTN gateways are not using H.323 or SIP, Mobile Voice Access functionality can still be provided using hairpinning on a separate gateway running H.323. Mobile Voice Access using hairpinning relies on off-loading the VoiceXML functionality to a separate H.323 gateway. Figure 21-24 illustrates a Mobile Voice Access call flow using hairpinning. In this example, just as in the previous example, the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 408-555-2345 (step 1). The call comes into the enterprise PSTN gateway (step 2) and is forwarded to Unified CM for call handling (step 3). Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 972 555 3456 as the number they wish to reach (followed by the # sign).
Note When using Mobile Voice Access with hairpinning, users calling into the system will not be identified automatically by their caller ID. Instead, users will have to key in their remote destination number manually prior to entering their PIN. The reason the user is not automatically identified is that, for hairpinning deployments, the PSTN gateway must first route the call to Unified CM to reach the hairpinned Mobile Voice Access gateway. Because the call is routed to Unified CM first, the conversion of the calling number from a mobile number to an enterprise directory number occurs prior to the call being handled by the Mobile Voice Access gateway. This results in the Mobile Voice Access gateway being unable to match the calling number with a configured remote destination, and therefore the system prompts the user to enter their remote destination number. This is unique to hairpinning deployments; with normal Mobile Voice Access flows, the PSTN gateway does not have to route the call to Unified CM first in order to access Mobile Voice Access because the functionality is available on the local gateway.
In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). The outbound call to 972 555-3456 is routed through the PSTN gateway (step 7). Finally, the call rings at the PSTN destination phone with number 972 555-3456 (step 8).
Figure 21-24 Mobile Voice Access Using Hairpinning
Note When deploying Mobile Voice Access in hairpinning mode, Cisco recommends configuring the Mobile Voice Access DID at the PSTN gateway and the Mobile Voice Access Directory Number within Cisco Unified CM (under Media Resources > Mobile Voice Access) as different numbers. A translation pattern within Unified CM can then be used to translate the called number of the Mobile Voice Access DID to the configured Mobile Voice Access directory number. Because the Mobile Voice Access directory number configured within Unified CM is visible to the administrator only, translation between the DID and directory number will be invisible to the end user and there will be no change in end-user dialing behavior. This is recommended in order to prevent mobility call routing issues in multi-cluster environments. This recommendation does not apply to Mobile Voice Access in non-hairpinning mode.
Note Mobile Voice Access in hairpinning mode is supported only with H.323 VXML gateways.
Enterprise Feature Access with Two-Stage Dialing Functionality
Figure 21-25 illustrates the call flow for Enterprise Feature Access two-stage dialing. In this example, the mobility user at remote destination phone 408 555-7890 dials the Enterprise Feature Access DID 408 555-2345 (step 1). Once the call is connected, the remote destination phone is used to send DTMF digits to Unified CM via the PSTN gateway, beginning with the user's PIN (followed by the # sign) which is authenticated with Unified CM. Next a 1 (followed by the # sign) is sent to indicate a two-stage dialed call is being attempted, followed by the phone number the user wishes to reach. In this case the user enters 9 1 972 555 3456 as the destination number (step 2).
Note Unlike with Mobile Voice Access, Enterprise Feature Access requires that all two-stage dialed calls must originate from a phone that has been configured as a remote destination in order to match the caller ID and PIN against the end-user account. There is no provision within Enterprise Feature Access in which the mobility user can enter their remote destination number or ID to identify themselves to the system. Identity can be established only via the combination of incoming caller ID and entered PIN.
Next the outgoing call is originated via the user's remote destination profile (step 3), and the call to PSTN number 972 555-3456 is routed via the enterprise PSTN gateway (step 4). Finally, the call rings the PSTN phone (step 5: in this case, 972 555-3456). As with Mobile Voice Access, the voice media path of each Enterprise Feature Access two-stage dialed call is hairpinned within the enterprise PSTN gateway utilizing two gateway ports.
Figure 21-25 Enterprise Feature Access Two-Stage Dialing Feature
Note In order for Enterprise Feature Access two-stage dialing to work as in Figure 21-25, ensure that the system-wide Enable Enterprise Feature Access service parameter is set to True.
Desk and Remote Destination Phone Pickup
Because Mobile Voice Access and Enterprise Feature Access functionality is tightly integrated with the Single Number Reach feature, once a Mobile Voice Access or Enterprise Feature Access two-stage dialed call has been established, the user does have the option of using Single Number Reach functionality to pick up the in-progress call on their desk phone by simply hanging up the call on the originating phone and pushing the Resume softkey on their desk phone or by using the mid-call hold feature. In turn, the call can then be picked up on the user's configured remote destination phone by pressing the Mobility softkey and selecting Send Call to Mobile Phone.
Enabling and Disabling Single Number Reach
In addition to providing users of Mobile Voice Access and Enterprise Feature Access with the ability to make calls from the PSTN as though they are within the enterprise, the functionality provided by Mobile Voice Access on the H.323 or SIP VoiceXML gateway and provided by Enterprise Feature Access also gives users the ability to remotely enable and disable their Single Number Reach functionality for each remote destination via their phone keypad. Rather than entering a 1 to make a call, users enter a 2 to turn the Single Number Reach feature on and a 3 to turn the Single Number Reach feature off.
If a user has more than one remote destination configured when using Mobile Voice Access, they are prompted to key in the remote destination phone number for which they wish to enable or disable the Single Number Reach feature. When using Enterprise Feature Access, a user can enable or disable Single Number Reach only for the remote destination phone from which they are calling.
Note When the Enable Mobile Voice Access service parameter is set to False, resulting in an inability to make two-stage dialed calls, Mobile Voice Access still provides users with the ability to enable and disable Single Number Reach remotely. As long as the Mobile Voice Access Directory Number has been configured on the system, the user's account has been enabled for Mobile Voice Access, and the Cisco Unified Mobile Voice Access service is running on the publisher, an authorized calling user can still enable or disable Single Number Reach.
Mobile Voice Access and Enterprise Feature Access Number Blocking
Administrators might want to prevent users of Mobile Voice Access and Enterprise Feature Access two-stage dialing from dialing certain numbers when using these features. In order to restrict or block calls to certain numbers when using these features for off-net calls, a comma-separated list of those numbers can be configured in the System Remote Access Blocked Numbers service parameter field. Once this parameter is configured with blocked numbers, those numbers will not be reachable from a user's remote destination phone when using Mobile Voice Access or Enterprise Feature Access features. Numbers that administrators might want to block can include emergency numbers such as 911. When configuring blocked numbers, ensure they are configured as they would be dialed by an enterprise user, with appropriate prefixes or steering digits. For example, if an emergency number is to be blocked and the emergency number is dialed by system users as 9911, then the number configured in the System Remote Access Blocked Numbers field should be 9911.
Access Numbers for Mobile Voice Access
While the Unified CM system allows the configuration of only a single Mobile Voice Access Directory Number, this does not preclude the use of multiple externally facing numbers that can access these internally configured numbers. For example, consider a system deployed in the US in New York with a remote site in San Jose as well as an overseas site in London. Even though the system may have the Mobile Voice Access directory number configured as 555-1234, the gateways at each location can be configured to map a local or toll-free DID number to this Mobile Voice Access directory number. For example, the gateway in New York may have DIDs of +1 212 555 1234 and +1 800 555 1234, which both map to the Mobile Voice Access number, while the gateway in San Jose has a DID of +1 408 666 5678 and the gateway in London has a DID of +44 208 777 0987, which also map to the Mobile Voice Access number of the system.
By acquiring multiple local or toll-free DID numbers, system administrators can ensure that Mobile Voice Access two-stage dialed calls will always originate as a call into the system that is either local or toll-free, thus providing further reductions in telephony costs.
Remote Destination Configuration and Caller ID Matching
When authenticating users for Mobile Voice Access and Enterprise Feature Access two-stage dialing functionality as well as the DTMF-based mid-call features Transfer and Conference, the caller ID of the calling remote destination phone is matched against all remote destinations configured within the system. Matching of this caller ID depends on a number of factors, including how the remote destination numbers are configured, whether digit prefixing is required to include PSTN steering digits on the system, and whether the Matching Caller ID with Remote Destination parameter is set to Partial or Complete Match. In all cases, the requirement is to be able to uniquely identify each mobility user based on the their remote destination number or numbers. For this reason, it is critical not only that remote destination numbers be configured uniquely within the system, but also that inbound caller ID matching (whether using complete or partial matching) must always uniquely correspond to a single remote destination. If a single or unique match is not found, caller ID matching will fail.
To control the nature of this matching, consider the following two approaches.
Using Complete Caller ID Matching
With this approach, remote destination numbers are configured exactly as the caller ID would be presented from the PSTN. For example, if the caller ID from the PSTN for a remote destination phone is presented to the system as 4085557890, then this number should be configured on the Remote Destination configuration page.
In order to route Single Number Reach calls appropriately to this remote destination, it is necessary to configure the dial plan to use either +E.164 dialing methods or a digit prefix mechanism to prefix necessary PSTN access codes and other required digits. For example, if you are not using a global +E.164 dial plan and assuming a 9 or other PSTN steering digits or country codes are required to reach the PSTN when dialing calls from the enterprise, then digit prefixing must be configured to add the appropriate PSTN steering digit and country code to the beginning of the configured remote destination number. Digit prefixing should be facilitated by using translation patterns, route patterns, or route list constructs within the Unified CM system. When using this complete match approach and a digit prefixing method, the Matching Caller ID with Remote Destination parameter should be left at the default setting of Complete Match.
Application Dial Rules may also be used to provide digit prefixing in these scenarios. However, it is worth noting that Application Dial Rules are applied based on called digit-string length and cannot be partitioned, meaning that they are applied globally across the system. This severely limits the use of Application Dial Rules, especially in scenarios where multiple dialing domains (for example, different countries) need to be supported on a single Unified CM cluster.
Note Not only are Application Dial Rules applied to Single Number Reach, Mobile Voice Access, and Enterprise Feature Access calls, but they are also applied to calls made with Cisco WebDialer, Cisco Unified CM Assistant, and Cisco Jabber applications. For this reason, exercise care when configuring these rules to ensure that dialing behavior across all applications is as expected.
The recommended dial plan approach is always to globalize the caller ID to +E.164 on ingress from the PSTN and always to configure remote destinations as +E.164. This will guarantee that the caller ID from the PSTN (after normalization) will always provide a unique match when compared against all configured remote destinations. Combined with a dial plan supporting +E.164 dialing, this eliminates the need for digit prefixing and ensures unique identification of remote destination users and numbers even when supporting multiple international numbering plans. Because the recommended dial plan approach is to globalize the caller ID on ingress and localize on egress according to trunk requirements and/or user expectations, using the unmodified caller ID as presented from the PSTN is not compatible with this approach.
Using Partial Caller ID Matching
With this approach, remote destinations are configured as they would be dialed from the system to the PSTN. For example, if the number for the remote destination is 14085557890 and PSTN access from the system requires a 9, then this number should be configured on the Remote Destination configuration page as 914085557890. This approach precludes the need for configuration of a digit prefixing mechanism on the system, but it requires setting the Matching Caller ID with Remote Destination service parameter to Partial Match and setting the Number of Digits for Caller ID Partial Match to the appropriate number of consecutive digits that should be matched against the remote destination caller ID. For example, if the caller ID for a remote destination is 14085557890 and the remote destination is configured as 914085557890, then the Number of Digits for Caller ID Partial Match would ideally be set to 10 or 11. In this example, this parameter could be set to a lower number of digits; however, always ensure that enough consecutive digits are matched so that all configured remote destinations in the system are matched uniquely. If there is no exact match or if more than one configured remote destination number is matched when using partial caller ID matching, the system treats this as if there is no matching remote destination number, thus requiring the user to enter their remote destination number/ID manually in the case of Mobile Voice Access before providing their PIN. With Enterprise Feature Access, there is no mechanism for the user to enter their remote destination number; therefore, when using this functionality, ensure that only unique matches occur.
Note If the PSTN service provider sends variable-length caller IDs, using partial caller ID matching is not recommended because ensuring a unique caller ID match for each inbound call might not be possible. In these scenarios, using complete caller ID matching and/or a +E.164 dial plan is the preferred method.
Mobile Voice Access and Enterprise Feature Access Architecture
The architecture of the Mobile Voice Access and Enterprise Feature Access feature is as important to understand as their functionality. Figure 21-26 depicts the message flows and architecture required for Mobile Voice Access and Enterprise Feature Access. The following sequence of interactions and events can occur between Unified CM, the PSTN gateway, and the H.323 or SIP VXML gateway:
1. Unified CM forwards IVR prompts and instructions to the H.323 or SIP VXML gateway via HTTP (see step 1 in Figure 21-26). This provides the VXML gateway with the ability to play these prompts for the inbound Mobile Voice Access callers.
2. The H.323 or SIP VXML gateway uses HTTP to forward Mobile Voice Access user input back to Unified CM (see step 2 in Figure 21-26).
3. The PSTN gateway forwards DTMF digits in response to user or Smart Phone key sequences from the remote destination phone for Enterprise Feature Access two-stage dialing and mid-call features (see step 3 in Figure 21-26).
Figure 21-26 Mobile Voice Access and Enterprise Feature Access Architecture
Note While Figure 21-26 depicts the H.323 or SIP VoiceXML gateway as a separate box from the PSTN gateway, this is not an architectural requirement. Both VoiceXML functionality and PSTN gateway functionality can be handled by the same box, provided there are no requirements for the PSTN gateway to run a protocol other than H.323 or SIP. An H.323 or SIP gateway is required for Mobile Voice Access VoiceXML functionality.
Note Because Cisco IOS XE does not provide native VoiceXML support, the Cisco 4000 Series ISR cannot be used as the VoiceXML gateway for Mobile Voice Access. If the PSTN gateway is a Cisco 4000 Series ISR, you most offload the VoiceXML functionality to a Cisco IOS gateway with native VoiceXML support.
High Availability for Mobile Voice Access and Enterprise Feature Access
The Mobile Voice Access and Enterprise Feature Access features rely on the same components and redundancy mechanisms as the Single Number Reach feature (see High Availability for Single Number Reach). Unified CM Groups are necessary for PSTN gateway registration redundancy. Likewise, PSTN physical gateway and gateway connectivity redundancy should be provided. Redundant access between the PSTN and the enterprise is required for remote destination phones to access Mobile Voice Access and Enterprise Feature Access features in the event of a gateway failure. However, while physical redundancy can and should be provided for the H.323 or SIP VoiceXML gateway, there is no redundancy mechanism for the Cisco Unified Mobile Voice Access service on Unified CM. This service can be enabled and run on the publisher node only. Therefore, if the publisher node fails, Mobile Voice Access functionality will be unavailable. Enterprise Feature Access and two-stage dialing functionality have no such dependency on the publisher and can therefore provide equivalent functionality to mobility users (without the IVR prompts).
Designing Cisco Unified Mobility Deployments
The Cisco Unified Mobility solution delivers mobility functionality via Cisco Unified CM. Functionality includes Single Number Reach, Mobile Voice Access, and Enterprise Feature Access. When deploying this functionality it is important to understand dial plan implications, guidelines and restrictions, and performance and capacity considerations.
Dial Plan Considerations for Cisco Unified Mobility
In order to configure and provision Unified Mobility appropriately, it is important to understand the call routing behavior and dial plan implications of the remote destination profile configuration.
Remote Destination Profile Configuration
When configuring Unified Mobility, you must consider the following two settings on the Remote Destination Profile configuration page:
This setting combines with the directory number or line-level calling search space (CSS) to determine which partitions can be accessed for mobility dialed calls. This affects calls made by the mobility user from the remote destination phone, including Mobile Voice Access and Enterprise Feature Access two-stage dialing as well as calls made in conjunction with mid-call transfer and conferencing features. Ensure that this CSS, in combination with the line-level CSS, contains all partitions that need to be accessed for enterprise calls originating from a user's remote destination phone. In a +E.164 dial plan using the line-only traditional approach with local route groups, this CSS is not required and can be set to <None>.
- Rerouting Calling Search Space
This setting determines which partitions are accessed when calls are sent to a user's remote destination phone. This applies to all Single Number Reach calls. When a call to a user's enterprise directory number is also sent via Single Number Reach to a user's remote destination, this CSS determines how the system reaches the remote destination phone. For this reason, the CSS should provide access to partitions with appropriate route patterns and gateways for reaching the PSTN or mobile voice network.
When configuring the Remote Destination Profile Rerouting CSS, Cisco recommends that the route patterns within this CSS point to a gateway that is in the same call admission control location as the gateway used to route the inbound call to the user's desk phone. This ensures that a call admission control denial due to insufficient bandwidth between two locations will not occur when routing calls out to the remote destination. Further, because subsequent call admission control checks after the initial Single Number Reach call is routed will not result in a denial if there is insufficient WAN bandwidth, routing the inbound and outbound call legs out a gateway or gateways in the same call admission control location ensures that subsequent desk phone or remote destination pickup operations during this call will not require call admission control, which could result in WAN bandwidth oversubscription.
When using route patterns pointing to route lists that use Standard Local Route Group, the local route group configured on the caller’s device pool will be used. In this case the egress gateway for the call leg to the remote destination will be local to the original calling device. For calls coming in from the PSTN, this will help to fulfill the above requirement to use egress gateways in the same call admission control location as the original caller (in this case the incoming gateway).
Likewise, it is equally important to ensure that call admission control denials are minimized when placing two-stage dialed calls. Call admission control denials for two-stage dialed calls can be minimized or avoided by using local route group constructs so that the egress gateway used to route the outbound call leg is chosen by the ingress gateway of the inbound call leg. With this method, the ingress and egress gateways used will be in the same call admission control location. Alternatively, the route patterns within the Remote Destination Profile device-level CCS should point to an egress gateway that is in the same call admission control location as the ingress gateway that handled the inbound call leg to the Mobile Voice Access or Enterprise Feature Access system access number. However, be aware that a subsequent desk phone pickup can result in WAN bandwidth oversubscription if the desk phone is in a different call admission control location than the gateway through which the Mobile Voice Access or Enterprise Feature Access system access numbers are reached.
Automatic Caller ID Matching and Enterprise Call Anchoring
Another aspect of the Unified Mobility dial plan that is important to understand is the system behavior with regard to automatic caller ID identification for inbound calls from configured remote destination phones. Whenever an inbound call comes into the system, the presented caller ID for that call is compared against all configured remote destination phones. If a match is found, the call will automatically be anchored in the enterprise, thus allowing the user to invoke mid-call features and to pick up in-progress calls at their desk phone. This behavior occurs for all inbound calls from any mobility user's remote destination phone, even if the inbound call is not originated as a mobility call using Mobile Voice Access or Enterprise Feature Access.
Note Automatic inbound caller ID matching for configured remote destination numbers is affected by whether the Matching Caller ID with Remote Destination service parameter is set to Partial or Complete Match. See Remote Destination Configuration and Caller ID Matching, for more information about this setting.
In addition to automatic enterprise call anchoring, inbound and outbound call routing must also be considered when a configured remote destination phone is calling into the enterprise. Inbound call routing for calls from configured remote destinations occurs in one of two ways, depending on the setting of the service parameter Inbound Calling Search Space for Remote Destination. By default, this service parameter is set to Trunk or Gateway Inbound Calling Search Space. With the service parameter set to the default value, inbound calls from configured remote destinations will be routed using the Inbound Calling Search Space (CSS) of the PSTN gateway or trunk on which the call is coming in. If, on the other hand, the parameter Inbound Calling Search Space for Remote Destination is set to the value Remote Destination Profile + Line Calling Search Space, inbound calls coming from remote destinations will bypass the Inbound CSS of the PSTN gateway or trunk and will instead be routed using the associated Remote Destination Profile CSS (in combination with the line-level CSS).
Given the nature of inbound call routing from remote destination phones, it is important to make sure that calling search spaces are configured appropriately in order to provide access for these inbound calls to any partitions required for reaching internal enterprise phones, thus ensuring proper call routing from remote destination phones.
Note Incoming calls that do not come from a configured remote destination phone are not affected by the Inbound Calling Search Space for Remote Destination service parameter because they will always use the trunk or gateway inbound CSS.
Outbound call routing for Mobile Voice Access or Enterprise Feature Access calls always uses a concatenation of the Remote Destination Profile line CSS and device-level CSS, therefore it is important to make sure that these calling search spaces are configured appropriately in order to provide access to any route patterns necessary for off-net or PSTN access, thus ensuring proper outbound call routing from remote destination phones.
Intelligent Session Control and Ring All Shared Lines
The Intelligent Session Control feature enables automatic call anchoring for enterprise-originated calls made directly to configured remote destination numbers. Normally, mobility call anchoring is dependent exclusively on calls made to or on behalf of a user's enterprise number. The system already anchors externally originated calls made by enterprise two-stage dialing because these call are routed as internal calls. With the Intelligent Session Control feature enabled, the system will also anchor internally originated calls made directly to configured remote destinations.
This feature is enabled by setting the Reroute Remote Destination Calls to Enterprise Number service parameter to True. By default, this service parameter is set to False and the feature is disabled. When the feature is enabled, not only will the system route the call to the dialed remote destination by way of the PSTN, but it will also automatically anchor the call inside the enterprise gateway. By anchoring these types of calls, the system enables the called mobile user to invoke mid-call features and desk phone pickup or session handoff.
As an example, assume that the Intelligent Session Control feature has been enabled and that a mobility-enabled user has a remote destination number configured as 408 555 1234, which corresponds to their mobile number. If another system user dials the mobility-enabled user's remote destination number (408 555 1234) from their desk phone, the system will route the call through the PSTN to the remote destination and will simultaneously anchor the call in the enterprise gateway. Once the call is set up and anchored, the called mobility-enabled user now has the ability to invoke mid-call features such as hold, transfer, and conference, as well as the ability to perform a desk phone pickup or session handoff.
Taking this same example and assuming instead that the Intelligent Session Control feature is disabled, then when a system user dials the mobility-enabled user’s remote destination directly from a desk phone inside the enterprise, the call will still be routed to the called remote destination through the PSTN; however, the call will not be anchored. As a result, the mobile user would not be able to invoke mid-call hold or transfer and would have no ability to perform a desk phone pickup or session handoff.
When enabling this feature, it is important to understand the implications to dial plan configuration and call routing. To invoke the feature, the number dialed by an internal user to reach a remote destination number on the PSTN (including any required PSTN steering digits) must match the remote destination (or mobility identity) number as it is configured on the system. For example, if the remote destination number is configured on the system as 408 555 1234 but internal users must normally dial PSTN steering digits 91 in addition to the number they are calling, then rerouting and resulting enterprise call anchoring will not occur. This is because the user dialed 91 408 555 1234 to reach the remote destination on the PSTN but the remote destination was configured as 408 555 1234, so there is no match.
For this feature to function properly, matching must occur between the configured remote destination and the number that must be dialed to reach this remote destination on the PSTN. To ensure that this matching happens, set the service parameter Matching Caller ID with Remote Destination to Partial Match. By setting this parameter to Partial Match and then specifying the number of digits to partially match using the Number of Digits for Caller ID Partial Match service parameter, it is still possible to match the configured remote destination number with the dialed number even if it contains PSTN steering digits.
Using the previous example and assuming that system has been set to use partial match on ten digits, the dialed number 9 1 408 555 1234 can be matched to the configured remote destination 408 555 1234. This is because, with partial matching, the system attempts to match the same number of digits as specified by the Number of Digits for Caller ID Partial Match, which in this case is ten digits. The system attempts to match the two numbers by matching digits from right to left. The last ten digits of the dialed number 9 1 408 555 1234 are 408 555 1234, and these ten digits match the ten digits of the configured remote destination (408 555 1234). In this example, the resulting call is anchored in the enterprise and the called mobile user is able to invoke mid-call features and perform desk phone pickup or session handoff.
At first glance it might appear that an easier way to handle this feature would be to configure remote destination or mobility identity numbers that include any required PSTN steering digits. However, when configuring these numbers with required PSTN steering digits, if you do not also configure partial caller ID matching, the system will not be able to perform automatic caller ID matching and enterprise anchoring for inbound calls from configured remote destinations or mobility identities. In the previous example, if the remote destination number had been configured as 9 1 408 555 1234 and complete caller ID matching had been used, an inbound call from the remote destination would present caller ID of 408 555 1234 and a match would not occur, meaning the inbound call from the remote destination would not be anchored as expected.
Based on this potential for mismatch between dialed numbers for outbound calls and configured remote destination numbers for inbound calls, Cisco recommends enabling partial (rather than complete) caller ID matching when using the Intelligent Session Control feature for all deployments that require one or more steering digits to reach the PSTN. This ensures that calls made directly to the remote destination number using PSTN steering digits are still matched and anchored. On the other hand, if steering digits are not required to reach the PSTN and users are able to dial the full E.164 number to route calls to the PSTN, then Cisco recommends the complete caller ID matching setting because the remote destination is configured to match the caller ID and is the same number as dialed by internal users to reach the remote destination or mobility identity on the PSTN.
When enabling the Intelligent Session Control feature, it is also important to understand the behavior of the enterprise and remote destination lines during the reroute feature operation. On call reroute, remote destination line settings Do Not Disturb (DND), Access Lists and Time of Day call filtering, and the Delay Before Ringing Timer are ignored. All reroute calls are routed unfiltered and immediately. Enterprise desk phone line settings are also ignored or bypassed by default. However, Call Forward All settings on the enterprise desk phone line can be honored during reroute feature operation by setting the Ignore Call Forward All on Enterprise DN service parameter to False. If this parameter is set to False, on reroute operation, calls will not be routed to the remote destination if the enterprise desk phone line has a call-forward-all destination set. Instead, the call will be routed to the call-forward-all destination. By default, this service parameter is set to True, and call-forward-all settings on enterprise desk phone lines are ignored.
Intelligent Session Control functionality may be further enhanced by using the Ring All Shared Lines feature. This feature is enabled by setting the Ring All Shared Lines service parameter to True. By default, this service parameter is set to True and the feature is enabled. However, the Ring All Shared Lines feature is dependent on the Intelligent Session Control feature, which must also be enabled in order use the Ring All Shared Line functionality. When the Ring All Shared Lines and Intelligent Session Control features are both enabled, not only will the system route internally originated calls to the dialed remote destination by way of the PSTN, but all of the user's other shared-line devices will also receive the call. This includes the user's enterprise desk phone as well as other configured remote destinations. The called user will then be able to answer the incoming call on any of their devices and the call will be anchored in the enterprise.
Note If Ring All Shared Lines is enabled, mobile client devices will not receive calls at the cellular voice interface of the device when the device is registered to Unified CM.
Caller ID Transformations
Any calls made into the cluster by configured remote destination numbers will automatically have their caller ID or calling number changed from the calling remote destination phone number to the enterprise directory number of the associated desk phone. For example, if a remote destination phone with number 408 555-7890 has been configured and associated to a user's enterprise desk phone with number 555-1234, then any call from the user's remote destination phone destined for any directory number in the cluster will automatically have the caller ID changed from the remote destination number of 408 555-7890 to the enterprise directory number of 555-1234. This ensures that the active call caller ID display and call history log caller ID reflect a mobility user's enterprise desk phone number rather than their mobile phone number, and it ensures that any return calls are made to the user's enterprise number, thus anchoring those calls within the enterprise.
Likewise, calls from a remote destination phone to external PSTN destinations and anchored in the enterprise via Mobile Voice Access or Enterprise Feature Access two-stage dialing, or those calls forked to the PSTN as a result of Single Number Reach, will also have caller ID changed from the calling remote destination phone number to the associated enterprise directory number.
Finally, in order to deliver the calling party number as an enterprise DID number rather than an enterprise directory number to external PSTN phones, calling party transformation patterns can be used. By using calling party transformation patterns to transform caller IDs from enterprise directory numbers to enterprise DIDs, return calls from external destinations will be anchored within the enterprise because they will be dialed using the full enterprise DID number. For more information about these transformations and dial plan implications, see Special Considerations for Cisco Unified Mobility.
Intelligent Proximity for Mobile Voice and Unified Mobility Interactions
The Intelligent Proximity for Mobile Voice feature on the Cisco DX Series endpoints and Cisco IP Phones 8851 and 8861 is compatible with the Unified Mobility feature set, including single number reach (SNR), remote destination and desk phone pickup, two-stage enterprise dialing, and mobile voicemail avoidance. For more information on Intelligent Proximity for Mobile Voice and Bluetooth pairing on the DX Series endpoints and 8851 and 8861 IP Phones, see Intelligent Proximity.
Guidelines and Restrictions for Unified Mobility
Note The Cisco Unified Mobility solution is verified with only Cisco equipment. This solution may also work with other third-party PSTN gateways and Session Border Controllers (SBCs), but each Cisco Mobility feature is not guaranteed to work as expected. If you are using this solution with third-party PSTN gateways or SBCs, Cisco technical support may not be able to resolve problems that you encounter.
The following guidelines and restrictions apply with regard to deployment and operation of Single Number Reach within the Unified CM telephony environment:
- Single Number Reach is supported only with PRI TDM PSTN connections. T1 or E1-CAS, FXO, FXS, and BRI PSTN connections are not supported. This PRI requirement is based on the fact that Cisco Unified CM must receive expeditious answer and disconnect indication from the PSTN in order to ensure full feature support. Answer indication is needed in order for Cisco Unified CM to stop ringing the desk phone and other remote destinations when a Single Number Reach call is answered at a particular remote destination. In addition, answer indication is required in order to support the single enterprise voicemail box feature. Finally, disconnect indication is required for desk phone pickup. A PRI PSTN connection will always provide answer or disconnect indication.
- Single Number Reach is also supported over SIP trunk VoIP PSTN connections. Use of Cisco IOS Unified Border Element is recommended as the demarcation point between the Unified CM SIP trunk and the service provider trunk. A VoIP-based PSTN connection is still able to provide expeditious answer and disconnect indication to Unified CM due to the end-to-end signaling path provided by VoIP-based PSTN connections.
- Single Number Reach can support up to two simultaneous calls per user. Any additional calls that come in are automatically transferred to the user’s voicemail.
- Single Number Reach does not work with Multilevel Precedence and Preemption (MLPP). If a call is preempted with MLPP, Single Number Reach features are disabled for that call.
- Single Number Reach services do not extend to video calls. A video call received at the desktop phone cannot be picked up on the cellular phone.
- Remote destinations must be Time Division Multiplex (TDM) devices or off-system IP phones on other clusters or systems. You cannot configure IP phones within the same Unified CM cluster as remote destinations.
- Mobile Voice Access VoiceXML capabilities are not supported with Cisco IOS XE software. Because there is no native VoiceXML support with Cisco IOS XE, the Cisco 4000 Series ISR cannot serve as a VXML gateway for Mobile Voice Access. Instead, deploy a separate H.323 Cisco IOS gateway for VoiceXML capabilities and configure Mobile Voice Access for hairpinning.
For additional guidelines and restrictions, refer to the information on Cisco Unified Mobility in the latest version of the Feature Configuration Guide for Cisco Unified Communications Manager, available at
Capacity Planning for Cisco Unified Mobility
Cisco Unified Mobility supports a maximum of 40,000 remote destinations or mobility identities per Unified CM cluster. The maximum number of mobility-enabled users would thus be 40,000 users, assuming a single remote destination or mobility identity per user. As the number of remote destinations or mobility identities per user increases, the number of supported mobility-enabled users decreases.
Note A mobility-enabled user is defined as a user that has a remote destination profile and at least one remote destination or a mobile client device and a mobility identity configured.
Note A mobility identity is configured just like a remote destination within the system, and it has the same capacity implications as a remote destination. Unlike a remote destination, however, the mobility identity is associated directly to a phone device rather than a remote destination profile. The mobility identity applies only to dual-mode mobile client devices running Cisco Jabber.
Scalability and performance of Cisco Unified Mobility ultimately depends on the number of mobility users, the number of remote destinations or mobility identities each user has, and the busy hour call attempt (BHCA) rates of those users. Multiple remote destinations per user and/or high BHCA per user can result in lower capacity for Cisco Unified Mobility. For more information on Cisco Unified Mobility sizing, including Unified CM server node capacities and hardware specific per-node and per-cluster capacities, see the chapter on Collaboration Solution Sizing Guidance.
Design Considerations for Cisco Unified Mobility
Observe the following design recommendations when deploying Unified Mobility:
- Ensure that the PSTN gateway protocol is capable of out-of-band DTMF relay or allocate media termination points (MTPs) in order to covert in-band DTMF to out-of-band DTMF. When using Cisco IOS gateways for PSTN connectivity, out-of-band DTMF relay will be supported. However, third-party gateways might not support a common out-of-band DTMF method, and as a result an MTP might be required. In order to use Enterprise Feature Access Two-Stage Dialing and mid-call features, DTMF digits must be received out-of-band by Cisco Unified CM.
Note When relying on MTP for converting in-band DTMF to out-of-band DTMF, be sure to provide sufficient MTP capacity. If heavy or frequent use of Enterprise Feature Access Two-Stage Dialing or mid-call features is anticipated, Cisco recommends a hardware-based MTP or Cisco IOS software-based MTP.
- Prior to deploying Unified Mobility, it is important to work with the PSTN provider to ensure the following:
– Caller ID is provided by the service provider for all inbound calls to the enterprise. This is a requirement if Enterprise Feature Access Two-Stage Dialing or mid-call transfer, conference, and directed call park features are needed.
– Outbound caller ID is not restricted by the service provider. This is a requirement if there is an expectation that mobility-enabled users will receive the caller ID of the original caller at their remote destination rather than a general enterprise system number or other non-meaningful caller ID.
Note Some providers restrict outbound caller ID on a trunk to only those DIDs handled by that trunk. For this reason, a second PRI trunk that does not restrict caller ID might have to be acquired from the provider. To obtain an unrestricted PRI trunk, some providers might require a signed agreement from the customer indicating they will not send or make calls to emergency numbers over this trunk.
Note Some providers allow unrestricted outbound caller ID on a trunk as long as the Redirected Dialed Number Identification Service (RDNIS) field or SIP Diversion Header contains a DID handled by the trunk. The RDNIS or SIP Diversion Header for forked calls to remote destinations can be populated with the enterprise number of the user by checking the Redirecting Number IE Delivery - Outbound check box on the gateway or trunk configuration page. Contact your service provider to determine if they honor the RDNIS or SIP Diversion Header and allow unrestricted outbound caller ID.
- Because mobility call flows typically involve multiple PSTN call legs, planning and allocation of PSTN gateway resources is extremely important for Unified Mobility. In cases where there are large numbers of mobility-enabled users, PSTN gateway resources will have to be increased. The following methods are recommend to minimize or reduce PSTN utilization:
– Limit the number of remote destinations per mobility-enabled user to one (1). This will reduce the number of DS0s that are needed to extend the inbound call to the user's remote destination. One DS0 is consumed for each configured remote destination when a call comes into the user's enterprise directory number, even if the call is not answered at one of the remote destinations. Note that a DS0 per remote destination may be used for as long as 10 seconds, even if the call is not answered at the remote destination.
– Use access lists to block or restrict the extension of calls to a particular remote destination based on incoming caller ID. Because access lists can be invoked based on the time of day, this eliminates the need for repeated updates of access lists by the end-user or the administrator.
– Educate end-users to disable Single Number Reach when not needed, to further eliminate DS0 utilization when a call comes in for that user's enterprise number. If Single Number Reach is disabled, incoming calls will still ring the desk phone and will still forward to enterprise voicemail if the call goes unanswered.
- Due to the potential for call admission control denials resulting from insufficient WAN bandwidth between locations and the possibility that a desk phone pickup or remote destination pickup might result in WAN bandwidth over-subscription, Cisco recommends configuring Remote Destination Profile CSS and Rerouting CSS so that route patterns within these CSSs point to gateways that are located within the same call admission control location as the gateway on which the inbound call leg comes in. For more information, see Remote Destination Profile Configuration.
- If you enable the Intelligent Session Control feature in deployments where PSTN steering digits must be dialed to access the PSTN, Cisco recommends setting the Matching Caller ID with Remote Destination service parameter to Partial Match and configuring the appropriate number of digits (Number of Digits for Caller ID Partial Match service parameter) to achieve a partial match of configured remote destinations or mobility identities. This will ensure proper functioning of the Intelligent Session Control feature and the mobility automatic caller ID matching and anchoring features.
Cisco Mobile Clients and Devices
As the prevalence of mobile users, mobile phones, and mobile carrier services continues to increase, the ability to use a single device for voice, video, and data services both inside and outside the enterprise becomes increasingly attractive. Mobile devices, including dual-mode smartphones and the clients that run on them, afford an enterprise the ability to provide customized voice, video, and data services to users while inside the enterprise and to leverage the mobile carrier network as an alternate connection method for general voice and data services. By enabling voice, video, and data services inside the enterprise and providing network connectivity for mobile client devices, enterprises are able to provide these services locally or remotely at reduced connectivity costs. For example, voice over IP (VoIP) calls made on the enterprise network will typically incur less cost than those same calls made over the mobile voice network.
In addition to providing voice and video over IP (VVoIP) capabilities, these mobile clients and devices enable mobile users to access and leverage other back-end collaboration applications and services. Other services and applications that can be leveraged through Cisco mobile clients and services include enterprise directory, enterprise voicemail, and XMPP-based enterprise IM (instant messaging) and presence. Further, these clients and devices can be deployed in conjunction with Cisco Unified Mobility so that users can leverage additional features and functions with their mobile device, such as Single Number Reach, enterprise two-stage dialing through Mobile Voice Access or Enterprise Feature Access, and single enterprise voicemail box.
This section examines mobile client architecture and common functions and features provided by Cisco mobile clients and devices, including remote secure attachment and handoff considerations related to moving an active voice call between the enterprise WLAN network and the mobile voice network. After covering the general mobile client solution architecture and features and functions, this section provides coverage of various capabilities and integration considerations for the following specific mobile clients and devices:
- Cisco Jabber — A mobile client available for Android and Apple iOS mobile devices, including iPhone and iPad, providing the ability to make voice and/or video calls over IP on the enterprise WLAN network or over the mobile data network as well as the ability to access the corporate directory and enterprise voicemail services and XMPP-based enterprise IM and presence.
- Cisco Spark — A mobile client available for Android and Apple iOS devices, including iPhone and iPad, providing 1-to-1 and 1-to-many cloud-based collaboration rooms enabling voice and/or video calls over IP, secure persistent messaging, and file sharing.
- Cisco WebEx Meetings — A mobile client available for Android, BlackBerry, Windows Mobile, and Apple iOS devices including iPhone and iPad, enabling users to attend and participate in Cisco WebEx meetings while mobile.
- Cisco AnyConnect Mobile — A mobile client available for Android and Apple iOS devices, enabling secure remote VPN connectivity to the enterprise for access to on-premises collaboration applications and services even when the user is outside of the enterprise.
In addition, this section discusses high availability and capacity planning considerations for Cisco mobile clients and devices.
Cisco Mobile Clients and Devices Architecture
Cisco mobile clients are deployed on a wide range of mobile devices including tablets and handheld devices with only IP-based network connectivity capabilities (IEEE 802.11 wireless local area network or mobile provider data network) and dual-mode phones, which contain two physical interfaces or radios that enable the device to connect to both mobile voice and data carrier networks by means of traditional cellular or mobile network technologies and to connect to wireless local area networks (WLANs) using 802.11. Cisco mobile clients and devices enable on-premises data and real-time traffic (voice and video) connectivity through an 802.11 WLAN. In addition, these clients and devices provide remote data and real-time traffic (voice and video) connectivity to the enterprise through public or private WLANs or over the mobile data network. For devices with provider cellular voice radios, voice connectivity may also be enabled through the mobile voice network and PSTN.
Note The use of the term dual-mode phone in this section refers specifically to devices with 802.11 radios in addition to the cellular radio for carrier voice and data network connectivity. Dual-mode devices that provide Digital Enhanced Cordless Telecommunications (DECT) or other wireless radios and/or multiple cellular radios are outside the scope of this section.
Figure 21-27 depicts the basic Cisco mobile clients and devices solution architecture for connecting and enabling mobile client devices for Cisco Collaboration deployments. For voice and video services, mobile client devices associate to the enterprise WLAN or connect over the Internet (from a public or private WLAN hot spot or the mobile data network), and the Cisco mobile client registers to Cisco Unified CM as an enterprise phone using the Session Initiation Protocol (SIP). Once registered, the client device relies on the underlying enterprise Cisco IP telephony network for making and receiving calls. When the mobile device is connected to the enterprise network and the client is registered to Unified CM, the device is reachable through the user's enterprise number. Any inbound calls to the user's enterprise number will ring the mobile client device. If the user has a Cisco IP desk phone, then the mobile client registration enables a shared line instance for the user's enterprise number so that an incoming call rings both the user's desk phone and the mobile device. When unregistered, the mobile client device will not receive incoming enterprise calls unless the mobile device has an active cellular voice radio, the user has been enabled for Cisco Unified Mobility, and Single Number Reach has been turned on for the user's mobile phone number. In these scenarios the mobile voice network and PSTN are used for making and receiving voice-only calls.
Unified Mobility features such as Single Number Reach are not compatible with tablets and other mobile client devices that do not have cellular voice radios because these non-dual-mode devices do not have a native PSTN reachable number. Non-dual-mode devices are able to make and receive enterprise calls only when connected to the enterprise and registered to the enterprise call control system.
As shown in Figure 21-27, when attached to the enterprise, Cisco mobile clients and devices can also communicate directly with other back-end application servers such as the corporate directory, Cisco Unity Connection enterprise voicemail system, and the Cisco IM and Presence Service for access to additional enterprise collaboration services such as messaging and presence. Cisco mobile clients and devices also integrate with cloud-based collaboration services such as Cisco WebEx, which delivers IM and presence and web conferencing services.
Figure 21-27 Cisco Mobile Clients and Devices Architecture
Note The voice and video quality of calls will vary depending on the Wi-Fi or mobile data network connection. Cisco Technical Assistance Center (TAC) is not able to troubleshoot connectivity or voice and video quality issues over 3G/4G mobile data networks or non-corporate Wi-Fi networks.
Dual-mode mobile client devices must be capable of dual transfer mode (DTM) in order to be connected simultaneously to both the mobile voice and data network and the WLAN network. This allows the device to be reachable and able to make and receive calls on both the cellular radio and WLAN interface of the device. In some cases proper mobile client operation might not be possible if mobile voice and data networks do not support dual-connected devices.
Voice and Video over Wireless LAN Network Infrastructure
Before considering the various mobile client device features and functions and the impact these features and functions have on the enterprise telephony infrastructure, it is critical to plan and deploy a finely tuned, QoS-enabled, and highly available WLAN network. Because dual-mode phones and other mobile devices rely on the underlying WLAN infrastructure for carrying both critical signaling and other traffic for setting up calls and accessing various applications as well as the real-time voice and video media traffic, deploying a WLAN network optimized for both data and real-time media traffic is necessary. A poorly deployed WLAN network will be subjected to large amounts of interference and diminished capacity, leading not only to poor voice and video quality but in some cases dropped or missed calls. This will in turn render the WLAN deployment unusable for making and receiving calls. Therefore, when deploying dual-mode phones and other mobile devices, it is imperative to conduct a WLAN radio frequency (RF) site survey before, during, and after the deployment to determine appropriate cell boundaries, configuration and feature settings, capacity, and redundancy to ensure a successful deployment of voice and video over WLAN. Each mobile device type and/or client should be tested on the WLAN deployment to ensure proper integration and operation prior to a production deployment. Using a WLAN that has been deployed and configured to provide optimized real-time traffic over WLAN services (such as the Cisco Unified Wireless Network), including quality of service, will ensure a successful mobile client device deployment.
Cisco recommends relying on the 5 GHz WLAN band (802.11a/n/ac) whenever possible for connecting mobile clients and devices capable of generating voice and video traffic. 5 GHz WLANs provide better throughput and less interference for voice and video calls.
For more information on voice and video over WLAN deployments and wireless device roaming, see Wireless Device Roaming.
Note While dual-mode phones and other mobile client devices are capable of connecting back to the enterprise through the Internet for call control and other Unified Communications services, Cisco cannot guarantee voice and video quality or troubleshoot connectivity or voice and video quality issues for these types of connections. These types of connections include remote connections to the enterprise through public or private WLAN access points (APs) or hot spots or through the mobile data network. Cisco recommends an enterprise class voice and video-optimized WLAN network for connecting dual-mode phones and other mobile client devices. Most public and private WLAN APs and hot spots are tuned for data applications and devices. In these cases, the AP radios are turned to maximum power, and dynamic-power control results in devices enabling maximum power on network attachment, which allows for larger client capacities. While this may be ideal for data applications that are capable of retransmitting dropped or lost packets, for real-time traffic applications this can result in poor voice and video quality due to the potential for large numbers of dropped packets. Likewise, mobile provider data networks are susceptible to congestion and/or dropped connections, which can also result in poor call quality and dropped calls.
Cloud or Off-Premises Collaboration Infrastructure
Cisco WebEx and Cisco Spark, cloud services available from the Cisco, do not require any hardware to be deployed on the enterprise premises. All services (audio, video, messaging, file and content sharing, and meeting and collaboration room information) are securely hosted in the Internet or the cloud. This means that all the content, voice, and video traffic from every client traverses the internet and is mixed and managed in the Cisco Collaboration Cloud.
The Cisco Collaboration Cloud infrastructure provides WebEx and Cisco Spark capabilities to mobile clients and devices, including:
- WebEx Meetings, which provides web-enabled voice and video conferencing with content sharing.
- WebEx Messenger, which provides XMPP IM and presence as well as point-to-point audio and video calling.
- Cisco Spark, which provides 1-to-1 and 1-to-many collaboration rooms with voice and video calling, messaging, and file sharing.
Mobile Client and Device Quality of Service
Cisco mobile client applications and devices generally mark Layer 3 QoS packet values in accordance with Cisco collaboration QoS marking recommendations. Table 21-3 summarizes these markings.
Table 21-3 Cisco Mobile Client Layer 3 QoS Markings
Voice media (audio only)
Video media (audio and video)
Cisco mobile client Layer 2 802.11 WLAN packet marking (User Priority, or UP) presents challenges given the various mobile platform and firmware restrictions. Because Cisco mobile clients run on a variety of mobile devices, Layer 2 wireless QoS marking is inconsistent and therefore Layer 2 wireless QoS marking cannot be relied on to provide appropriate treatment to traffic on the WLAN.
Despite appropriate mobile client application Layer 3 or even Layer 2 packet marking, mobile devices present many of the same challenges as desktop PCs in terms of generating many different types of traffic, including both data and real-time traffic. Given this, mobile devices generally fall into the untrusted category of collaboration endpoints. For deployments where mobile client devices are not considered trusted endpoints, packet marking or re-marking based on traffic type and port numbers is required to ensure that network priority queuing and dedicated bandwidth is applied to appropriate traffic. In addition to re-marking the mobile device traffic, Cisco recommends using network-based policing and rate limiting to ensure that the mobile client devices do not consume too much network bandwidth.
Alternatively, given appropriate Cisco mobile client Layer 3 marking and assuming mobile client devices are trusted, Cisco mobile client traffic will be queued appropriately as it traverses the enterprise network by using priority voice queuing and dedicated video media and call signaling bandwidth queues.
Cisco Mobile Clients and Devices Features and Functions
Cisco mobile clients and devices provide a range of features and functions. While features and operations may vary from device to device, the common operations and behaviors described in this section apply to all non-cloud-based Cisco mobile clients.
Enterprise Call Routing
Because Cisco mobile clients and devices are capable of making and receiving calls using the enterprise telephony infrastructure and call control services, it is important to understand the nature and behavior of call routing as it pertains to mobile client devices.
Inbound Call Routing
When mobile clients and devices register to Unified CM as an enterprise device with enterprise number, the mobile device rings when incoming calls to the system are destined for the user's enterprise number. This occurs for incoming calls originated on the PSTN or from other Unified CM clusters or enterprise IP telephony systems as well as for incoming calls originated within the Unified CM cluster by other users. If the mobile client device user has other devices or clients that are also associated to the enterprise number, these devices will also ring as shared lines; and once the call is answered at one of the devices or clients, ringing of all other devices and clients ceases.
In scenarios where a user has been enabled for Cisco Unified Mobility, and when Single Number Reach is enabled for the user's dual-mode mobile phone number, the incoming call may also be extended to the mobility identity corresponding to the user's mobile phone number. However, this depends on whether the mobile device is connected to the enterprise WLAN network or attached to the enterprise network through a secure connection and registered to Unified CM. In situations in which the device is connected to the enterprise network directly or through a secure remote connection, an incoming call to the user's enterprise number will not be extended by Single Number Reach to the mobility identity of the mobile device even if Single Number Reach is enabled on for this mobile number. The reason an incoming call to the enterprise number is not extended to the mobility identity of a dual-mode mobile device when it is registered to Unified CM is that the system is aware the device is connected to the enterprise network and available. Thus, in order to reduce utilization of enterprise PSTN resources, Unified CM does not extend the call to the dual-mode mobile phone's mobile voice network interface through the PSTN. Instead, only the WLAN or mobile data network interface corresponding to the enterprise number receives the call.
Note In cases where dial via office is enabled (see Dial Via Office), even if the client is registered, Unified CM will extend inbound calls to the user's mobile number using Single Number Reach rather than via VoIP to the enterprise number.
For situations in which the mobile device is not connected to the enterprise network directly or through a secure remote connection or is not registered to Unified CM, incoming calls to the enterprise number will be extended to the dual-mode mobile phone number per the configured mobility identity, assuming that the user has been enabled for Unified Mobility and that Single Number Reach for the mobility identity is turned on. For more information on integration of mobile clients and devices with Unified Mobility, see Interactions Between Cisco Jabber and Cisco Unified Mobility.
The same behavior and logic described above also applies with the Ring All Shared Lines feature. If this feature is enabled, calls are extended to the mobility identity or cellular number only when the dual-mode mobile client device is not registered to Unified CM. For more information on the Ring All Share Line feature, see Intelligent Session Control and Ring All Shared Lines.
In all cases, incoming calls made directly to the dual-mode device's mobile network phone number will always be routed directly to the mobile voice interface of the dual-mode device on the mobile network, unless the provider network or device settings are such that calls are not extended to the device by the mobile network. This is considered appropriate behavior because these calls were not made to the user's enterprise number. These would be considered personal calls, and as such should not be routed through the enterprise.
Note Mobile client devices that do not have cellular voice radios, such as tablet devices, are not dual-mode devices and as such cannot be reached on a mobile voice network interface. These devices can be reached only at the enterprise number by voice-over-IP.
Outbound Call Routing
For outbound calls from the dual-mode mobile device, the interface used depends on the location and connectivity of the device at that particular time. If the dual-mode device is not connected to the enterprise and not registered to Unified CM, then calls are routed by the cellular voice radio interface to the mobile voice network as usual. However, when connected to the enterprise and registered to Unified CM, the mobile device should make all calls through the enterprise telephony infrastructure. If no enterprise connectivity is available or the mobile client is unregistered, then outbound calling is not possible from the enterprise number, and instead calls would have to use the mobile number of the mobile client device for making calls over the mobile voice network. Alternatively, users may use the two-stage dialing features provided with Cisco Unified Mobility (see Mobile Voice Access and Enterprise Feature Access).
The enterprise dial plan determines the dialing behavior of the mobile client device when it is connected to the enterprise and registered to Unified CM. For example, if the enterprise dial plan is configured to allow abbreviated dialing to reach internal extensions, then a mobile device registered to Unified CM can use this abbreviated dialing. While it is certainly a convenience for dual-mode mobile phone users to be able to dial within the enterprise using enterprise dialing habits and abbreviated dialing as well as site-based and/or PSTN steering digits for outbound calls, it is also a somewhat unnatural dialing scheme because mobile phone users typically dial numbers for outgoing calls on their mobile phone by using full E.164 dial strings since this is what is expected by the mobile voice network for outbound calling.
The enterprise dialing experience for an end-user is ultimately up to the enterprise policies and administrator of the enterprise telephony deployment. However, for dual-mode mobile devices, Cisco recommends normalizing required dialing strings for dual-mode client devices so that user dialing habits are maintained whether the device is connected to the enterprise network and registered to Unified CM or not. Because dialing on the mobile voice network is typically done using full +E.164 (with a preceding '+') and mobile phone contacts are typically stored with full +E.164 numbers, Cisco recommends configuring the enterprise dial plan to accommodate full +E.164 with preceding '+' for dual-mode mobile devices. When the dial plan is configured within Unified CM to handle this type of outbound dialing for dual-mode phones, it is possible for users to store a single set of contacts on the phone in the +E.164 format and, when dialed from these contacts or manually using the full +E.164 number, calls will always be routed to the appropriate destination, whether the device is connected to the enterprise network directly or over secure remote connection and registered to Unified CM or connected only to the mobile voice network. Configuring the enterprise dial plan in this manner provides the best possible end-user dialing experience so that users' mobile device dialing habits are maintained and they do not have to be aware of whether the device has enterprise connectivity and is registered to Unified CM.
To achieve normalized dialing from dual-mode phones, whether connected to the enterprise or just the mobile voice network, configure the dial plan within Unified CM with the following considerations in mind:
- Ensure that the enterprise dial plan is capable of handling dial strings from dual-mode phones typically used on the mobile voice network. For example, the dial plan should be configured to handle the following strings, which might be dialed from a mobile phone to reach a particular phone through the mobile voice network: +1 408 555 1234, 408 555 1234. Supporting the latter 10-digit dialing method (for example, 408 555 1234) might potentially overlap with other dialing habits such as abbreviated intra-site dialing. In that case the administrator has to decide which of the colliding dialing habits (10-digit dialing or abbreviated intra-site) should be available for dual-mode phones registered to the enterprise network. The set of dialing habits supported on dual-mode phones often differs from the set of dialing habits supported on regular endpoints.
- For calls destined for other enterprise numbers, systems configured for abbreviated dialing should be capable of modifying dial strings and rerouting to enterprise extensions as appropriate. For example, assuming the enterprise dial plan is based on five-digit internal dialing, the system should be configured to handle call routing to an enterprise extension so that a call to made to +1 408 555 1234 or 408 555 1234 is modified and rerouted to 51234 if the call is made while the dual-mode device is registered to Unified CM.
- Ensure that all inbound calls to the enterprise destined for dual-mode devices have the calling number and/or caller ID prefixed with appropriate digits so that missed, placed, and received call history lists are in full +E.164 formats. This will allow dual-mode device users to dial from call history lists without the need for editing the dial string. Instead, users will be able to select a number from the call history list to redial, whether connected to the enterprise or not. For example, if an incoming call from inside the enterprise originates from 51234 to a dual-mode user's enterprise number and the call goes unanswered, Unified CM should be configured to manipulate the calling number so that the resulting entry within the history list of the dual-mode device shows either 408 555 1234 or +1 408 555 1234. This number can be dialed whether the dual-mode device is connected to the enterprise or just to the mobile voice network without the need for further manipulation.
The one exception to normalized dialing for dual-mode devices is for scenarios in which some enterprise extensions or phones are reachable only internally (that is, they have no externally reachable corresponding DID number). In these situations, non-externally reachable numbers can be dialed (manually or from contacts) using abbreviated formats. Because these numbers will never be available externally and can be dialed only from inside the enterprise, some sort of enterprise-only indication should be made when storing these numbers in the contact list. Further, incoming calls from these internal-only numbers should not have the calling number modified for call history lists because these numbers may be called only inside the enterprise. Instead, calls from these extensions should be listed in all call history lists without modification so that the abbreviated dial strings can be successfully dialed only while the device is connected to the enterprise and registered to Unified CM.
Mobile client devices that do not have cellular voice radios, such as tablets, are dependent exclusively on enterprise connectivity and enterprise voice and video telephony or cloud-based collaboration services.
Emergency Services and Dialing Considerations
Mobile client devices do present a slight challenge when it comes to making calls to emergency service numbers such as 911, 999, and 112. Because the mobile client devices may be located inside or outside the enterprise, providing location indication of a device and its user in the event of an emergency must be considered. Dual-mode mobile devices with cellular voice radios receive location services from their provider networks, and these location services are always available when the device is connected and typically able to pinpoint locations far more precisely than enterprise wireless networks; therefore, Cisco recommends that dual-mode device users rely on the mobile voice network for making emergency calls and determining device and user location. To ensure that Cisco dual-mode client devices rely exclusively on the mobile provider voice network for emergency and location services, these clients force all calls made to numbers configured in the Emergency Numbers field on the mobile client device configuration page to route over the mobile voice network. Further, dual-mode phone users should be advised to make all emergency calls over the mobile voice network rather than the enterprise network.
While making emergency calls over WLANs or mobile data networks is not recommended, mobile devices that do not have cellular voice radios are capable of making calls only through these data interfaces. Mobile devices that do not have cellular voice radios should not be relied upon for making emergency calls.
Enterprise Caller ID
When mobile client devices are connected to the enterprise and registered to Unified CM (either through the mobile data network or a WLAN), all calls made with the enterprise line over the WLAN or mobile data network will be routed with the user's enterprise number as caller ID. This ensures that returned calls made from call history lists at the far-end are always routed through the enterprise because the return call is to the user's enterprise number. If a dual-mode mobile device user has been enabled for Cisco Unified Mobility, and Single Number Reach is turned on for the mobile phone number, return calls to the enterprise number would also be extended to the dual-mode device through the PSTN whenever it is not connected to the enterprise.
When mobile client devices are connected to the enterprise and registered to Unified CM as enterprise endpoints, they are able to invoke call processing supplementary services such as hold, resume, transfer, and conference, using SIP call signaling methods as supported by Unified CM. Just as with any IP phone or client registered to Unified CM, these devices are able to leverage enterprise media resources such as music on hold (MoH), conference bridges, media termination points, and transcoders.
External Call Routing
When dual-mode mobile client devices are not connected to the enterprise and/or not registered to Unified CM, they may make and receive calls only over the mobile voice network. For this reason, Unified CM has no visibility into any calls being made or received at the dual-mode mobile device while it is unregistered. The mobile number is the caller ID being sent to the network when calls are made from dual-mode phones not connected to the enterprise. This will likely result in unanswered calls being made directly back to the dual-mode device's mobile number instead of being routed back through the enterprise.
If the dual-mode mobile client device is integrated with Cisco Unified Mobility, enterprise two-stage dialing services may be leveraged for making calls through the enterprise network even when the dual-mode device is outside the enterprise and not registered to Unified CM. Unified Mobility two-stage dialing is done using either Mobile Voice Access or Enterprise Feature Access and requires the user to dial an enterprise system access DID number and enter credentials prior to dialing the number they are calling. For more information on Unified Mobility two-stage dialing features, see Mobile Voice Access and Enterprise Feature Access.
Likewise, if the dual-mode phone is integrated with Unified Mobility, a user can also receive incoming calls to their enterprise number at the mobile number through Single Number Reach; can invoke mid-call features using DTMF key sequences including hold, resume, transfer, and conference; and can perform desk phone pickup to move an active call from the mobile phone to the enterprise desk phone.
Remote Secure Enterprise Connectivity
Mobile client devices can utilize the IP telephony infrastructure for enterprise voice and video over IP calling and other collaboration services, even when not inside the enterprise, provided they have a secure connection back to the enterprise in order to register the client with Unified CM and to access other collaboration applications and services. Remote secure connectivity for these devices requires the use of the Cisco AnyConnect mobile client VPN solution or the VPN-less Cisco Expressway mobile and remote access feature in order to secure the client connection over the Internet.
Voice and video quality and user experience for remotely attached mobile client devices will vary depending on the nature of the Internet-based network connection. Cisco cannot guarantee acceptable voice and video quality nor successful connectivity for these types of client connections. Care should be taken when relying on these types of connections for business-critical communications. In the case of dual-mode devices with unreliable or low-bandwidth Internet connections, users with dual-mode devices should be advised to make calls over the mobile voice network if connectivity is available rather than relying on the remote enterprise telephony infrastructure.
Additional Services and Features
In addition to call processing or call control services, Cisco mobile clients and devices are capable of providing the additional features and services described in this section.
Dual-Mode Call Handoff
One very important aspect of dual-mode device deployments is call preservation as a user moves in and out of the enterprise or as the device connects to and disconnects from the enterprise network and network connectivity changes from the cellular voice radio to the WLAN radio, and vice versa. Because dual-mode phone users are often mobile, it is important to maintain any active call as a dual-mode user moves in or out of the enterprise. For this reason, dual-mode client devices and the underlying enterprise telephony network should be capable of some form of call handoff.
There are two types of call handoff that should be accommodated by both the dual-mode client and the underlying IP telephony infrastructure:
Call hand-out refers to the movement of an active call from the WLAN or mobile data network interface of the dual-mode phone to the cellular voice interface of the dual-mode phone. This requires the call to be handed out from the enterprise IP network to the mobile voice network through the enterprise PSTN gateway.
Call hand-in refers to the movement of an active call from the cellular voice interface of the dual-mode phone to the WLAN or mobile data network interface of the dual-mode phone. This requires the call to be handed in from the mobile voice network to the enterprise IP network through the enterprise PSTN gateway.
The handoff behavior of a dual-mode phone depends on the nature of the dual-mode client and its particular capabilities. Dual-mode client handoff may be invoked manually by the user or automatically based on network conditions. In manual handoff scenarios, the dual-mode users are responsible for engaging and completing the handoff operation based on their location and needs. With automatic handoff, the mobile client monitors the WLAN signal and makes handoff decision based on strengthening or weakening of the WLAN signal at the client. Hand-out is engaged in the case of a weakening WLAN signal, while hand-in is engaged in the case of a strengthening WLAN signal. Automatic handoff depends on the mobile device to provide capabilities for monitoring WLAN signal strength.
Handoff operations are critical for maximizing utilization of the enterprise IP telephony infrastructure for phone calls. These operations are also necessary for providing voice continuity and good user experience so that users do not have to hang up the original call and make another call to replace it.
XMPP-Based IM and Presence
Some mobile clients are capable of providing enterprise instant messaging (IM) and presence services based on the Extensible Messaging and Presence Protocol (XMPP), through integration to an on-premises or off-premises application server or service. In either case, the IM and presence capabilities of these mobile clients enable the following:
- Adding users to contact or buddy lists
- Setting and propagating user presence and availability status
- Reception of presence status for a buddy or contact
- Creating and sending of instant messaging (IM) or text messages
- Reception of IM or text messages
While IM and presence are not required functionality for mobile clients, they do enable users to make their availability status visible to contacts and to view the availability status of contacts, thus improving productivity. Further, users can send enterprise-based IM messages rather than incurring costs for mobile Short Message Service (SMS) messages.
Corporate Directory Access
Mobile clients and devices are capable of accessing the enterprise directory for contact lookups. Enterprise directory access is enabled using either:
- Lightweight Directory Access Protocol (LDAP) for communication between the clients and a compatible LDAP directory
- REST-based (HTTPS) communications between the clients and the User Data Services (UDS) API, which provides a set of operations that enable authenticated access to user contact information stored within the end-user database of the Unified CM cluster
Beginning with Cisco Unified CM 11.5, UDS-to-LDAP Proxy can also be used for contact searches. When enabled, contact searches are still handled by UDS but are proxied to the corporate LDAP directory, with UDS relaying results back to the mobile client. This enables mobile clients to search a corporate directory that exceeds the number of users supported within Unified CM.
Note Direct LDAP directory access is recommended over UDS directory access methods (local Unified CM database and UDS-to-LDAP proxy) due to the fact that UDS-based directory access greatly reduces Unified CM node endpoint capacity. However, beginning with Cisco Unified CM 11.5 and Jabber 11.5, UDS-based directory access no longer reduces endpoint capacity.
While corporate directory access is not a required feature for mobile devices and clients, it does provide a superior user experience for mobile users when they are able to access corporate directory information from their mobile device.
Enterprise Voicemail Services
Many mobile clients and devices are also capable of accessing enterprise voicemail services. Cisco mobile clients are capable of receiving enterprise message waiting indication whenever an unread voicemail is in the user's enterprise voicemail box and the mobile device is attached to the enterprise network. Further, mobile clients can be used to retrieve enterprise voicemail messages. Typically enterprise voicemail messages are retrieved when the user dials the voicemail system number and navigates to their voicemail box after providing required credentials. However, Cisco Jabber mobile clients provide the ability to retrieve voicemail messages from the voicemail box by downloading and displaying a list of all messages in the voicemail box and then by selecting individual messages to be downloaded to the mobile device for listening. This is sometimes referred to as visual voicemail. Both the mobile client and the enterprise voicemail system must be capable of providing and receiving message waiting indication (MWI), voicemail message information, and downloads of the messages over the network. Cisco Unity Connection supports visual voicemail through REST (HTTPS) and provides MWI, voicemail lists, and message downloads.
Dial Via Office
Dial via office (DVO) functionality provides automated enterprise dialing capabilities that enable dual-mode mobile devices to initiate calls through the enterprise telephony infrastructure. Deploying DVO calling provides the following benefits to the enterprise:
- Cost savings for calls to international and (possibly) long distance destinations as compared to direct-dialed cellular calls. Note that, in cases of mobile data traversal, mobile data costs must also be considered.
- Ability to dial internal enterprise numbers. Because DVO calls are made using the enterprise line, non-DID or internal-only enterprise extensions are reachable.
- Mobile phone number masking. For DVO calls, the system sends the user's enterprise number as caller ID, and not the mobile phone number.
- Centralized enterprise call detail records (CDRs) and call logs. Because DVO calls are made through the enterprise telephony infrastructure, administrators have complete visibility to these calls even though they traverse the PSTN and mobile voice network.
- Enterprise call anchoring. DVO calls are anchored in the enterprise, thus enabling users to leverage Cisco Unified Mobility DTMF-based mid-call features and desk phone pickup.
Dual-mode mobile devices running the Cisco Jabber client are able to make DVO calls using the Unified CM telephony infrastructure and enterprise PSTN gateway to make calls using the enterprise line. However, unlike voice over IP (VoIP) calling where voice media traverses the IP network, this functionality is facilitated by SIP signaling between the client and Unified CM over an IP connection (WLAN or mobile data) and voice media between the mobile device and the mobile voice network and PSTN, as shown in Figure 21-28.
Figure 21-28 Cisco Dial Via Office Architecture
Note For DVO calls, all voice or media from the user's mobile phone will always travel through the mobile voice network, PSTN, and enterprise PSTN gateway. Media never traverses the IP data connection to the enterprise. The mobile data network connection is used only for call signaling traffic and other application interactions.
For details on dial via office as implemented for Cisco Jabber clients, refer to Cisco Jabber Dial Via Office for Dual-Mode Devices.
Simplified Configuration for Mobile Client Users
Cisco mobile clients provides a streamlined configuration method for simplifying first-time end-user client configuration at the mobile client device. This configuration method relies on RFC 2782 standard Domain Name Service records (DNS SRV) within the corporate DNS server to automatically discover collaboration services on the network. DNS SRV records direct the mobile client to appropriate application servers for call control and IM and presence services. This configuration and provisioning method alleviates the need for the user to manually configure the XMPP IM and presence server and voice and video call control server or TFTP server host name or IP address. Instead the user simply enters their user ID and domain name, and the client application automatically discovers the available collaboration services and connects to these back-end servers, with the application prompting the user for credentials as appropriate. If no services are discovered or if service discovery operation fails, then the mobile client application reverts to manual configuration mode, requiring users to enter collaboration application server host names or IP addresses and credentials. Multiple DNS SRV records with priority and weighting indication ensure high availability of back-end collaboration application services as well as mobile client distribution across multiple servers providing these services.
Note Mobile client user simplified configuration does not simplify administrative tasks related to client and service configuration and provisioning on the back-end application servers. All administrative tasks to add user accounts, mobile client devices, and services configuration are still required in addition to creating the DNS SRV record or records in the corporate DNS server.
Cisco Bring Your Own Device (BYOD) Infrastructure
Cisco mobile client applications such as Cisco Jabber provide core Unified Communications and collaboration capabilities, including voice, video, and instant messaging to users of mobile devices such as Android and Apple iOS smartphones and tablets. When a Cisco mobile client device is attached to the corporate wireless LAN, the client can be deployed within the Cisco Bring Your Own Device (BYOD) infrastructure.
Because Cisco mobile clients and devices rely on enterprise wireless LAN connectivity or remote secure attachment through VPN or VPN-less connections, they can be deployed within the Cisco Unified Access network and can utilize the identification, security, and policy features and functions delivered by the BYOD infrastructure.
The Cisco BYOD infrastructure provides a range of access use cases or scenarios to address various device ownership and access requirements. The following high-level access use case models should be considered:
- Basic Access — This use case enables basic Internet-only access for guest devices. This use case provides the ability to enable employee-owned personal device network connectivity without providing access to corporate resources.
- Limited Access — This use case enables full access to corporate network resources, but it applies exclusively to corporate-owned devices.
- Enhanced Access — This use case enables granular access to corporate network resources for both corporate-owned devices and employee-owned personal devices based on corporate policies.
Cisco collaboration mobile clients, whether running on corporate or personal devices, usually require access to numerous back-end on-premises enterprise application components for full functionality. For this reason the Limited or Enhanced Access use case scenarios generally apply to applications such as Cisco Jabber for Android or iPhone. The chief difference between these two access models is that with Limited Access, the corporate-owned devices are given full access to corporate network resources. In the case of Enhance Access, not only is the scope expanded to include employee-owned devices, but access to corporate network resources can also be provided in a granular way so that devices and the applications that run on them are able to access only specific resources based on corporate security policies.
In the case of cloud-based collaboration services, Cisco mobile clients and devices connect directly to the cloud through the Internet without the need for enterprise network attachment. In these scenarios, user and mobile devices can be deployed using the Basic Access model because these use cases require only Internet access.
For more information about the Cisco BYOD infrastructure and BYOD access use cases, refer to the Cisco Bring Your Own Device (BYOD) Smart Solution Design Guide, available at
When deploying Cisco mobile clients and devices within the Cisco BYOD infrastructure, consider the following high-level design and deployment guidelines:
- The network administrator should strongly consider allowing voice and video-capable clients to attach to the enterprise network in the background (after initial provisioning), without user intervention, to ensure maximum use of the enterprises telephony infrastructure. Specifically, use of certificate-based identity and authentication helps facilitate an excellent user experience by minimizing network connection and authentication delay.
- In scenarios where Cisco mobile clients and devices are able to connect remotely to the enterprise network through a secure VPN or VPN-less connection:
– The network administrator should weigh the corporate security policy against the need for seamless secure connectivity without user intervention in order to maximize utilization of the enterprise telephony infrastructure. The use of certificate-based authentication and enforcement of a device pin-lock policy provides seamless attachment without user intervention and functionality similar to two-factor authentication because the end user must possess the device and know the pin- lock to access the network. If two-factor authentication is mandated, then user intervention will be required in order for the device to attach remotely to the enterprise.
– It is important for the infrastructure firewall configuration to allow all required client application network traffic to access the enterprise network. Failure to provide an appropriate access solution or to open access to appropriate ports and protocols at the corporate firewall could result in an inability of the Cisco mobile clients or devices to register to on-premises Cisco call control for voice and video telephony services and/or the loss of other client features such as enterprise directory access or enterprise visual voicemail.
- When enterprise collaboration applications such as Cisco Jabber are installed on employee-owned mobile devices, if the enterprise security policy requires the device to be wiped or reset to factory default settings under certain conditions, device owners should be made aware of the policy and encouraged to backup personal data from their device regularly.
- When deploying Cisco collaboration mobile clients and devices, it is important for the underlying network infrastructure from end-to-end to support the necessary QoS classes of service, including priority queuing for voice media and dedicated video and signaling bandwidth, to ensure the quality of client application voice and video calls and appropriate behavior of all features.
Deployment Considerations for Cisco Mobile Clients and Devices
This section discusses deployment considerations the following Cisco mobile clients and devices:
Cisco Jabber for Android and Apple iOS
This section describes characteristics and deployment considerations for Cisco Jabber.
Cisco Jabber mobile clients are available for Android and Apple iOS mobile devices, including iPad and iPhone. Once the client application is downloaded from the appropriate store or market (Apple Application Store or Google Play) and installed on the Apple iOS or Android device, it can connect to the enterprise network and register to Unified CM as a SIP enterprise phone.
To provide registration and call control services to the Cisco Jabber mobile client, the device must be configured within Unified CM as a Cisco Dual Mode for Android or iPhone, or Cisco Jabber for Tablet device type. Next, the mobile device must be configured to access the enterprise WLAN for connectivity based on the enterprise WLAN infrastructure and security policies. Alternatively the mobile device can be connected to the enterprise network via the mobile data network or over non-enterprise WLANs. Once the mobile device has been configured to access the enterprise network, when the Cisco Jabber client is launched, it will register the device to Unified CM. To integrate to Unified Mobility and to leverage handoff functionality, the mobile number of the Android or iPhone smartphone must be configured as a mobility identity associated to the Cisco Dual Mode for Android or iPhone device within Unified CM.
The Cisco Jabber client is supported on the following devices:
Various models of Android phones and tablets. (Consult the release notes referenced below for specific device and firmware support information.) These devices must be running a minimum firmware version of 4.1(2). Although not officially supported, Cisco Jabber for Android runs on many Android devices running version 4.1(2) or later, with various degrees of limitations depending on the device. The WLAN interfaces of most Android devices support 802.11a, 802.11b, 802.11g, 802.11n, and 802.11ac network connectivity.
Various Apple iOS devices including iPhone and iPad. (Consult the release notes referenced below for specific device and firmware support information.) These devices must be running a minimum iOS version of 7.1. The WLAN interfaces of most Apple iOS devices support 802.11a, 802.11b, 802.11g, and 802.11n network connectivity. Some newer Apple devices support 802.11ac.
For details on the latest specific device and firmware version support, refer to the product release notes for:
The Cisco Jabber for Android, iPad, and iPhone clients not only provide voice and video over IP phone services but also provide XMPP-based enterprise instant messaging (IM) and presence, corporate contact and directory services when configured to access the enterprise contact source, and enterprise voicemail message waiting indication (MWI) and visual voicemail when integrated to Cisco Unity Connection.
The Cisco Jabber clients running on smartphones (Android and iPhone) are capable of performing only manual hand-out as described in the section on Cisco Jabber Dual-Mode Handoff.
For more information about the Cisco Jabber Android and Apple iOS clients, additional feature details, and supported hardware and software versions, refer to the Cisco Jabber documentation for:
Cisco Jabber Service Discovery
As indicated previously, Cisco mobile clients such as Jabber are able to discover available collaboration services by relying on DNS lookups and DNS SRV service record resolution. When service discovery is properly configured, the user needs to enter only their user name and domain, and the client will automatically discover and connect to available collaboration services.
As shown in Figure 21-29, during initial client configuration or in the case of network connection changes, Jabber discovers collaboration services by querying DNS for the following SRV records:
- _cisco_uds._tcp. <domain>
SRV record or records of this type are added to the enterprise DNS server when Jabber is deployed in phone-only mode enabling voice and video over IP calling or in full UC mode enabling both voice and video calling well as IM and presence. If the query for this record is resolved by DNS, Cisco Jabber connects to Unified CM, determines the authenticator, and locates available services.
SRV record or records of this type are added to the enterprise DNS server when Jabber is deployed in IM-only mode enabling XMPP-based IM and presence. If the query for this record is resolved by DNS, Cisco Jabber connects to Unified CM IM and Presence and authenticates.
In the case of hybrid deployments with Cisco WebEx Messenger, during initial configuration and on network connection changes, the client also issues an HTTP query to a central authentication service (CAS) URL for Cisco WebEx Messenger service to determine if the domain is a valid WebEx domain. When the client receives positive confirmation to the HTTP query that a valid WebEx domain has been entered, the client then connects to and authenticates with the WebEx Messenger service and retrieves client configuration and information on available UC services as configured in the Cisco WebEx Org Admin.
Figure 21-29 Cisco Jabber Service Discovery
While the UDS service runs on all nodes in the Unified CM cluster, when configuring DNS SRV records for Unified CM UDS service, administrators should configure records for resolution to Unified CM subscriber nodes only. This ensures that client interaction with the UDS service avoids the publisher node and instead distributes the load across call processing nodes within the cluster.
In deployments where service discovery is not configured or reliance on DNS is not possible, the Jabber client will revert to manual configuration, requiring the user to enter authenticator and service node IP addresses. Manually configured IP addresses are cached by the Jabber client for use on subsequent connections.
Once service discovery or manual configuration is complete, Jabber must authenticate and download a service profile and/or the jabber-config.xml file (if available), which directs the client to additional back-end application services such as voicemail and directory and enables appropriate configuration.
Cisco Jabber Corporate Directory Access
Cisco Jabber mobile clients rely on various methods for accessing enterprise contact information. In addition to local device contacts and contacts previously added to the Jabber buddy list, Jabber mobile clients are also able to access corporate directory services using the following methods:
- Basic Directory Integration (BDI)
The BDI method of corporate directory access relies on LDAP communication between the Jabber client and supported LDAP compliant directories such as Microsoft Active Directory and OpenLDAP. BDI is the default method of directory integration for Jabber.
- Cisco Directory Integration (CDI)
The CDI method of corporate directory access was introduced with Jabber 11.8 and also relies on LDAP communication between the Jabber client and supported LDAP compliant directories such as Microsoft Active Directory and OpenLDAP. CDI is the default method of directory integration for Jabber 11.8 and later releases, and it replaces BDI.
- Unified CM User Data Services (UDS)
The UDS method of corporate directory access relies on HTTP communication between the Jabber client and Unified CM UDS services running on each Unified CM node.
- Unified CM UDS-to-LDAP Proxy
This method of corporate directory access relies on the Unified CM UDS service resolving or proxying directory searches against the corporate LDAP directory rather than using the local user directory. This method is available beginning with Cisco Unified CM 11.5. UDS-to-LDAP proxy allows Jabber users to search against the entire corporate directory rather than being limited by the local Unified CM cluster end-user database.
The jabber-config.xml file is used to configure the directory integration method for Jabber clients as well as to configure certain directory related settings for Jabber clients.
We recommend using the BDI method of directory access for on-premises clients due to performance limitations of Unified CM UDS. When directory integration is configured to use the UDS method, Unified CM node Jabber endpoint capacity is reduced by 50%. For example, when deploying a 5,000 user OVA Unified CM node and using the UDS method of directory access for Jabber clients, the Jabber endpoint capacity of the Unified CM node reduces from 5,000 to 2,500 Jabber devices (assuming no other endpoints are configured on the node).
Note Beginning with Cisco Unified CM 11.5 and Jabber 11.5, the use of UDS as a contact source no longer reduces system endpoint capacity.
When Jabber clients connect remotely using Expressway mobile and remote access, only UDS methods of directory access (local Unified CM database or UDS-to-LDAP proxy) are supported. Consider enabling UDS-to-LDAP proxy when the corporate directory size exceeds the local Unified CM directory size, to enable mobile client users to search the entire directory.
Cisco Jabber Dual-Mode Handoff
To properly deploy Cisco dual-mode clients such as Cisco Jabber, it is important to understand the nature of handoff operations within the client. The handoff method used by the Cisco Jabber dual-mode client depends on the Transfer to Mobile Network setting on the Cisco Dual Mode for iPhone or Cisco Dual Mode for Android device configuration page.
There are two methods of handoff, depending on the Transfer to Mobile Network setting:
With this method the Transfer to Mobile Network setting should be set to Use Mobility Softkey (user receives call). In this type of handoff, the Unified CM system generates a call over the PSTN to the user's mobile number.
With this method the Transfer to Mobile Network setting should be set to Use HandoffDN Feature (user places call). In this type of handoff, the mobile client generates a call over the mobile voice network to the handoff number configured within the Unified CM system.
Note Handoff capabilities apply only to dual-mode smartphones. This functionality is not supported on devices without cellular voice radios, such as the Samsung Galaxy Note Pro.
Mobility Softkey Method of Hand-Out
The operation depicted in Figure 21-30 is of an active call on an iPhone or Android dual-mode device within the enterprise being moved manually from the WLAN interface to the mobile voice network or cellular interface of the device through the enterprise PSTN gateway. As shown, there is an existing call between the mobile client device associated to the enterprise WLAN and registered to Unified CM, and a phone on the PSTN network (step 1). Because this is a manual process, the user must select the Use Mobile Network button from the in-call menu within the Cisco Jabber client, which signals to Unified CM the intention to hand-out the call (step 2). Next Unified CM generates a call to the configured mobility identity number corresponding to this mobile device through the enterprise PSTN gateway (step 3). This call to the mobility identity is made to the mobile voice network or cellular interface of the iPhone or Android device. The user can now move out of the enterprise and away from WLAN network coverage (step 4). In the meantime, the inbound call from Unified CM is received at the mobile voice network interface, and the user must answer the call manually to complete the hand-out. Once the inbound call on the cellular interface is answered, the RTP stream that was traversing the WLAN is redirected to the PSTN gateway, and the call continues uninterrupted between the mobile client device and the original PSTN phone, with the call anchored in the enterprise gateway (step 5).
Figure 21-30 Cisco Jabber Dual-Mode Hand-Out (WLAN-to-Mobile Voice Network): Mobility Softkey Method
Handoff Number Method of Hand-Out
Figure 21-31 illustrates the same hand-out operation as in Figure 21-30, where an active call on an iPhone dual-mode phone within the enterprise is moved manually from the WLAN interface to the mobile voice network or cellular interface of the device through the enterprise PSTN gateway. However, in this case the Handoff Number method of hand-out is used.
As shown in Figure 21-31, there is an existing call between the dual-mode device associated to the enterprise WLAN and registered to Unified CM, and a phone on the PSTN network (step 1). Because this is a manual process, the user must select the Use Mobile Network button from the in-call menu within the Cisco Jabber dual-mode client, which signals to Unified CM the intention to hand-out the call (step 2). Next the Cisco Jabber client automatically generates a call through the cellular interface over the mobile voice network to the configured Handoff Number within the Unified CM system (step 3). The user can now move out of the enterprise and away from WLAN network coverage (step 4). In the meantime, the inbound call from the Cisco Jabber client is received by Unified CM. Assuming the inbound calling number matches the user's configured mobility identity, the RTP stream that was traversing the WLAN is redirected to the PSTN gateway, and the call continues uninterrupted between the Cisco Jabber mobile client and the original PSTN phone, with the call anchored in the enterprise gateway (step 5).
Figure 21-31 Cisco Jabber Dual-Mode Hand-Out: Handoff Number Method
Note The Handoff Number method of hand-out requires Unified CM to receive an inbound calling number from the PSTN network that matches the mobility identity number configured under the Cisco Dual Mode device attempting the hand-out. If the caller ID is not sent by the dual-mode device, if the PSTN provider does not send the inbound caller ID to the enterprise, or if the inbound caller ID does not match the user's configured mobility identity, the hand-out operation will fail.
Note Cisco Jabber dual-mode clients do not support hand-in. In scenarios where an in-progress call is active between the dual-mode mobile voice network or cellular interface and an enterprise phone (or a PSTN phone with the call anchored in the enterprise gateway), the only way to move the call to the WLAN interface of the dual-mode device is to hang up the call and redial once the dual-mode client has connected to the enterprise network and registered to Unified CM.
WLAN Design Considerations for Cisco Jabber Mobile Clients
Consider the following WLAN guidelines when deploying Cisco Jabber mobile clients:
- Whenever possible, ensure that Cisco Jabber mobile clients roam on the WLAN only at Layer 2 so that the same IP address can be used on the WLAN interface of the device. In Layer 3 roaming scenarios where subnet boundaries are crossed due to device IP address changes, calls will be dropped.
- Deploy Cisco Jabber mobile clients on WLAN networks where the same SSID is used across all APs. Roaming between APs is much slower if SSIDs are different.
- Ensure all APs in the WLAN broadcast their SSID(s). If the SSID is not broadcast by the AP, the user may be prompted by the device to join other Wi-Fi networks or the device may automatically join other Wi-Fi networks. When this occurs the call is interrupted.
- Whenever possible, deploy Cisco Jabber mobile clients on the 5 GHz WLAN band (802.11a/n/ac). 5 GHz WLANs provide better throughput and less interference for voice and video calls.
Cisco Jabber Dial Via Office for Dual-Mode Devices
The Unified CM administrator can enable or disable dial via office (DVO) calling for each dual-mode device by using the Product Specific Configuration Layout section of the Cisco Dual Mode for iPhone or Android device configuration page. Once DVO is enabled, the user can turn on DVO using the Calling Options setting within the Cisco Jabber application. It is important to note that the DVO calling options dictate not only the outbound calling method used by the Jabber client but also the inbound calling method. Table 21-4 shows the various calling options and the corresponding outbound and inbound calling method based on the type of network connectivity.
Table 21-4 Inbound and Outbound Calling Method with Cisco Jabber Dial Via Office Calling Options
Cisco Jabber DVO Calling Options
802.11 WLAN (Corporate/enterprise)
Voice over IP
Voice over IP
Dial via office
Single Number Reach
Voice over IP
Voice over IP
802.11 WLAN (Non-corporate/enterprise)
Dial via office
Single Number Reach
Outbound call: Native cellular
Inbound call: Single Number Reach
The default calling option when DVO is first enabled is Autoselect, which results in voice over IP (VoIP) for both inbound and outbound Cisco Jabber calling when the device is connected over an 802.11 WLAN, while DVO will be used for outbound calling and Single Number Reach will be used for inbound calling when the device is connected over the mobile data network.
In all cases, calls made to emergency numbers configured in the Emergency Numbers field on the mobile client device configuration within Unified CM will be dialed directly over the cellular network regardless of the calling option selected.
Note The Dial via Office calling feature applies only to dual-mode smartphones. This functionality is not supported on tablets such as the Apple iPad because there is no cellular voice radio on those devices.
Note Support for the Dial via Office calling feature over Expressway mobile and remote access connections requires Cisco Unified CM 11.0(1a)SU1 or later and Expressway X8.7 or later releases.
When dial via office is enabled for Cisco Jabber clients, as with Single Number Reach, the mobile voicemail avoidance or single enterprise voicemail box feature of Cisco Unified Mobility is engaged. In the case of dial via office, this voicemail avoidance feature ensures that, given a failure in the network path or some other communication error during a DVO call setup, the called user does not end up in the calling user's voicemail box. Typically the User Control method of voicemail avoidance provides the best overall user experience because, if a DVO call leg inadvertently ends up being answered by a voicemail system, the call leg will be disconnected when a DTMF tone is not received by Unified CM, and the DVO call will be cleared. When Cisco Jabber users are enabled for the User Control method of mobile voicemail avoidance, they should be reminded that they must press a button on the mobile device key pad when receiving a mobility call at the client device. Failure to do so will result in call setup failure.
Note Because the User Control method of mobile voicemail avoidance is completely dependent on successful relay of the DTMF tone from the mobile device over the PSTN connection and PSTN gateway and out-of-band to Unified CM, failure to propagate inbound DTMF from the PSTN to Unified CM results in a disconnect of all enterprise calls made (dial via office reverse) or received (single number reach) by the mobile device. If DTMF cannot be effectively relayed from the PSTN to Unified CM, then the Timer Control mobile voicemail avoidance method should be used instead.
For more information about the single enterprise voicemail box voicemail avoidance feature, see Mobile Voicemail Avoidance with Single Enterprise Voicemail Box.
Dial Via Office Calling Option Use Cases
When deploying dial via office, consider the following Cisco Jabber client calling option user profiles:
The typical user profile for Autoselect is a user that is mobile both within and outside the office. For this user profile, Autoselect provides potential least cost routing by taking advantage of VoIP when 802.11 WLAN connectivity is available and falls back to mobile voice and data network (DVO and Single Number Reach) when WLAN connectivity is not available.
The typical user profile for the Mobile Voice Network calling option is a highly mobile user that almost never has WLAN coverage and whose mobile data connectivity does not provide acceptable throughput and reliability to ensure good voice quality and reliable calling over IP connections.
The typical user profile for the Voice over IP calling option is a user that is mobile within the office (home or enterprise) but for whom enterprise calling is not typically required outside the enterprise. Additionally, with this user profile, mobile voice and data costs are usually an important consideration for both corporate-paid and employee-paid mobile voice and data service.
Dial Via Office Reverse
Cisco Jabber clients support dial via office reverse (DVO-R). With this method of DVO, the call setup is facilitated by an inbound call from the Unified CM system to the user's configured mobility identity or mobile phone number.
Figure 21-32 illustrates a DVO-R call flow. In this example, a Cisco Jabber user wishes to dial a PSTN phone at +1 408 555-7890. The user dials the number or selects the number from the contact list from within the Cisco Jabber client, which generates a SIP call setup request over the IP connection to the enterprise and Unified CM (step 1). Based on the call setup request, Unified CM generates a reverse call back to the user's configured mobility identity (mobile phone number) using the enterprise PSTN gateway (step 2). Once the incoming call from Unified CM is answered at the mobile device, a call is extended to the number the user called or selected (step 3; in this case, +1 408-555-7890). Once the call is answered at the far end, the media path is connected and the call is anchored through the enterprise PSTN gateway (step 4). Because the call is now anchored in the enterprise gateway, the user has the ability at any point during this call to use the Unified Mobility desk phone pickup feature as well as to invoke Unified Mobility DTMF-based mid-call features.
Figure 21-32 Cisco Jabber Dial Via Office Reverse
Note The call flow shown in Figure 21-32 assumes that Cisco Jabber is registered to Unified CM, that DVO is enabled for the user, and that the client calling option setting is either Mobile Voice Network or Autoselect. If the client setting is Autoselect, the dual-mode device running Cisco Jabber must be IP-connected via the mobile data network. If connected over 802.11 WLAN, then the client would use voice over IP rather than DVO.
By default the DVO-R callback call leg will be extended to the user's mobile device, as shown in Figure 21-32; however; a user may specify an alternate callback number in the DVO Callback Number field within the Cisco Jabber client. By default the DVO Callback Number field is populated with the user's configured mobility identity. If the user configures a different number in this field, the DVO-R callback call leg will be extended to that number. For example, rather than receiving the callback on the mobile phone, the user may wish to direct the callback to their home phone.
Note When invoking DVO-R with an alternate callback number, if the callback call leg from Unified CM is directed to a user-specified alternate number, the call is not anchored in the enterprise. In such cases, users cannot perform desk phone pickup or invoke DTMF-based mid-call features on DVO-R calls using an alternate callback number. In addition, voicemail avoidance does not engage for DVO-R alternate number calls.
Mobile Profiles and Dial Via Office Reverse
Cisco Unified CM mobility profiles may be assigned to the mobility identity for mobile client devices. While not required, the mobility profile specifies the caller ID sent by the system during setup of the DVO-R callback call leg to the mobility identity or alternate callback number. The number configured in the Callback Caller ID field of the Dial-via-Office Reverse Callback Configuration section of the mobility profile configuration page is the number sent as caller ID. If no mobility profile is assigned to the mobility identity or if the Callback Caller ID field is left blank, the system will send the configured default Enterprise Feature Access Number.
Note The Mobile Client Calling Option field of the mobility profile has no impact on DVO operation; regardless of the setting, the Cisco Jabber client makes DVO-R calls when enabled for DVO calling. Dial via Office Forward (DVO-F) is not a currently available calling option.
Cisco Jabber Point-to-Point Calling
Cisco Jabber mobile clients are capable of providing point-to-point voice and video calling over IP without the need for Unified CM registration. Instead, the Jabber client leverages the Cisco WebEx Messenger cloud service for REST/HTTPS call signaling. Point-to-point call media leverages the RTP protocol with the G.722 codec for call audio and H.264 for call video. With REST point-to-point calling, only a single call per Jabber mobile client is supported, and mid-call supplementary features such as hold, resume, transfer, and conference are not supported.
IM Push Notifications for Cisco Jabber for iPhone and iPad
When running in the background on mobile devices, the Cisco Jabber client relies on a periodic keep-alive message to maintain connectivity to call control and IM and presence services so that incoming calls and messages are received and the user is notified. Beginning with Cisco Jabber for iPhone and iPad 11.8(1) and Cisco Unified CM and IM and Presence Service release 11.5 SU2 (or the latest version of WebEx Messenger), when the client is running in the background on an Apple iOS device, the client can receive IM notifications through the Apple Push Notification service (APNs). When client notification from the IM and Presence Service or WebEx Messenger is required, the IM and Presence Service or WebEx Messenger service sends a notification through the Cisco Collaboration Cloud, where it is relayed to APNs. In turn, APNs sends notification to the Apple iOS device, which has previously established trust with APNs during device activation. This notification through APNs triggers an alert to the user.
For on-premises IM and Presence deployments, APNs for Cisco Jabber for iPhone and iPad clients is enabled on Unified CM by the administrator. Once enabled, Cisco Jabber for iPhone and iPad clients running in the background will receive IM notifications through APNs.
Note Because Apple iOS 10 still supports keep-alive messages to maintain connectivity when the Cisco Jabber for iPhone and iPad client is running in the background, enabling APNs on Unified CM is not yet a requirement. However, once the current backgrounding capability is removed in a future Apple iOS release, APNs will be the only method for notifying users about an incoming IM when the Cisco Jabber for iPhone and iPad client is running in the background.
In the case of cloud or hybrid deployments using WebEx Messenger, APNs is enabled within the WebEx cloud by default, and Cisco Jabber for iPhone and iPad 11.8(1) and later clients will receive IM notifications through APNs while running in the background.
Jabber-to-Jabber calling with WebEx Messenger is not supported with APNs. If you plan to use the Jabber-to-Jabber calling feature with WebEx messenger, you will need to disable APNs manually with the < Policies > < Push_Notification_Enabled > parameter in the jabber-config.xml file. For more information on jabber-config.xml parameters, refer to the Parameters Reference Guide for Cisco Jabber, available at
When end-to-end encryption (AES) policy is set to enforced or optional with WebEx Messenger, APNs is automatically disabled and the client will receive IM notifications in the usual way when the client is running in the background.
Note The use of APNs for Jabber running in the background applies only to Cisco Jabber for iPhone and iPad clients. Windows, Mac, and Android Jabber clients are not impacted and will continue to receive notifications in the usual way when running in the background.
Cisco Jabber and Expressway Mobile and Remote Access
The mobile and remote access feature of the Cisco Expressway solution provides secure firewall traversal for Cisco Jabber, enabling remote Jabber users to access enterprise collaboration applications and services from their mobile devices when outside the enterprise.
All collaboration traffic traversing the Expressway mobile and remote access connection is encrypted, including call media and signaling. The encrypted connection is between the Jabber endpoint and the Expressway-C node inside the enterprise. Traffic between Expressway-C and endpoints or applications inside the enterprise is unencrypted by default. Media and signaling traffic inside the enterprise is encrypted only when the Unified CM cluster is configured as mixed mode with device authentication, SRTP media, and TLS SIP signaling encryption facilitated by security configuration relying on the Unified CM Cisco Certificate Trust List (CTL) Provider and Certificate Authority Proxy Function (CAPF) services.
Jabber determines its location relevant to the enterprise (inside or outside) based on DNS query resolution and a split DNS resolution design whereby the service records for Unified CM (_cisco-uds._tcp) and Unified CM IM and Presence (_cuplogin._tcp) are configured only in the corporate DNS and the service record for Expressway (_collab-edge._tls) is configured only on the public DNS. This split design ensures that corporate DNS resolution points Jabber directly to collaboration services when inside the enterprise and public DNS resolution points Jabber to connect through Expressway. DNS queries are sent by Jabber whenever the network connection of the mobile device changes.
As shown in Figure 21-33, Jabber queries DNS for three SRV service records: _cisco-uds._tcp, _cuplog._tcp, and _collab-edge._tls. When inside the enterprise, the Jabber client receives resolution from corporate DNS either pointing to Unified CM or Unified CM IM and Presence. In this case, Jabber will connect directly to the resolved collaboration application service node(s). When outside the enterprise, Jabber does not receive resolution for Unified CM or Unified CM IM and Presence from public DNS, but instead receives resolution for Expressway directing the client to connect to the enterprise through Expressway.
Figure 21-33 Cisco Jabber: Split DNS Resolution Inside and Outside the Enterprise
Note In cases where Cisco AnyConnect VPN is used for remote enterprise connectivity, Jabber will receive DNS query resolution from corporate DNS through the VPN tunnel and will connect directly to collaboration service nodes.
When deploying Expressway mobile and remote access for Cisco Jabber mobile clients, consider the following unsupported features and functions:
Moving an active call from the WLAN interface of the Jabber device to the cellular voice interface is not supported over Expressway connections.
- CAPF enrollment for endpoint authentication and media and signaling encryption
If secure media and signaling is required on the enterprise network, the Jabber device must complete CAPF enrollment while on-premises and prior to connecting over Expressway.
- Per-user or per-device access restrictions
There is no mechanism for restricting specific users or devices from connecting through Expressway mobile and remote access. If Expressway mobile and remote access is deployed and a user has been provisioned for Jabber on the collaboration infrastructure (Unified CM and Unified CM IM and Presence), then the user may connect through Expressway.
All calls and other collaboration application connections over Expressway mobile and remote access are cleared whenever the network path changes or is lost.
LDAP traffic is not enabled on Expressway mobile and remote access connections. For this reason all Jabber clients are forced to use a UDS method for corporate directory access when connecting over Expressway, even if the directory access method has been configured as BDI or CDI. As previously mentioned, prior to Unified CM 11.5 and Jabber 11.5 versions, Unified CM node Jabber endpoint capacity is reduced by 50% when directory access is facilitated by UDS. Given the reduced endpoint capacity for Jabber when connected through Expressway, additional Unified CM nodes might need to be deployed to handle the required capacity. Beginning with Unified CM 11.5 and Jabber 11.5, endpoint capacity is not reduced by UDS directory access.
If any of the above features and functions is required for the deployment, consider using AnyConnect VPN instead of Expressway for remote secure enterprise access.
Cisco Jabber and Expressway Mobile and Remote Access with Cisco AnyConnect VPN Split-Tunnel
In some cases VPN and Expressway might need to be deployed in parallel, enabling Jabber users to connect via either VPN or Expressway. In these situations, there are two methods of use. Jabber users can rely on the Expressway mobile and remote access feature for collaboration workloads and rely on VPN for all device traffic when connectivity back to the enterprise requires workloads outside of collaboration. In these scenarios, when the Cisco AnyConnect VPN client establishes a connection back to the enterprise, either due to VPN on-demand triggering or manual launch by the user, active connections are dropped and the user must wait for the Jabber client to reconnect to provisioned collaboration services over VPN before resuming use. This results in a poor user experience.
Alternatively, AnyConnect VPN and Expressway may be used simultaneously with split-tunneling to force collaboration flows through the Expressway mobile and remote access connection and all other traffic through the VPN tunnel. This alternative method often provides a better user experience because it prevents the Jabber client from disconnecting from Expressway and reconnecting over VPN when the VPN tunnel is established.
As shown in Figure 21-34, the split-tunneling afforded by this method of deployment relies on two basic principles
- DNS filtering at the Cisco Adaptive Security Appliance (ASA) VPN head-end
Traffic filtering at the ASA is used to filter DNS queries from the Jabber client for _cisco-uds._tcp. <domain> and _cuplogin._tcp. <domain>. Because these DNS queries are filtered, the Jabber client is unable to resolve Unified CM or IM and Presence service record requests for direct connection to collaboration services. Therefore, the only DNS resolution will be for _collab-edge._tcp. <domain>, which always results in Expressway connection and traversal.
- Exclusion of Expressway access over the VPN tunnel
IP address filtering at the ASA is used to prevent the Jabber client from connecting to the Expressway-E publicly facing interface. When filtering Expressway-E node public interface IP address(es), a split-tunnel VPN connection is created, resulting in Jabber traffic exclusion from the VPN tunnel and thus this traffic traverses Expressway while all other traffic traverses the VPN tunnel.
Figure 21-34 Cisco Jabber: Expressway Mobile and Remote Access and Cisco AnyConnect VPN
In the case of AnyConnect VPN split-tunneling with Expressway mobile and remote access, the same Expressway DNS SRV record (_collab-edge._tls) configured in the public DNS is added to the corporate DNS. This prevents the need to provide access and forward DNS queries to the public DNS through the VPN tunnel.
Although configuring an identical _collab-edge._tls SRV record in the corporate DNS would seem to violate the foundational split DNS design expected with Jabber and Expressway mobile and remote access deployments, in fact, Jabber's order of SRV resolution preference ensures appropriate behavior. Jabber's order of SRV resolution preference is for Unified CM (_cisco-uds._tcp) first, then IM and Presence (_cuplogin._tcp), and finally Expressway (_collab-edge._tls). Therefore, even when the _collab-edge._tls query can be resolved by the corporate DNS, the client will still connect directly to collaboration services because the corporate DNS will resolve queries for _cisco-uds._tcp or _cuplogin._tcp services first.
For more information about Jabber and Expressway mobile and remote access with AnyConnect VPN, refer to the information on mobile and remote access collaboration with Cisco Expressway Series, found in the Cisco Unified Access (UA) and Bring Your Own Device (BYOD) CVD available at
Cisco Jabber with SAML Single Sign-On
Cisco Jabber mobile clients are able to leverage single sign-on (SSO) using the Security Assertion Markup Language (SAML) version 2. Jabber and Cisco collaboration infrastructure including Unified CM, Unified CM IM and Presence, and Unity Connection leverage web-based SSO SAML v2 in order to identify and authenticate user connections, thus enabling the use of a single set of Jabber user credentials for access to all collaboration services.
As depicted in Figure 21-35, Cisco Jabber SSO depends on pre-established trust relationships between collaboration applications such as Unified CM, called service providers, and the identity provider (IdP). Unified CM and Unity Connection service providers rely on LDAP sync and integration with the corporate LDAP directory to identify users. Likewise, the IdP relies on the LDAP corporate directory for authentication of users. Supported IdPs for Cisco Jabber and collaboration services include Ping Federate, Microsoft Active Directory Federation Services (ADFS), and Open Access Manager (OpenAM).
Figure 21-35 shows a basic Jabber SSO flow. The SSO flow begins with the Jabber client requesting access to a collaboration service provider – for example, access to Unified CM for call control services. Rather than logging in directly to the collaboration service provider for access, the service provider redirects the Jabber client to the IdP with a SAML authentication request. The IdP requests authentication credentials from the Jabber user and authenticates the user against the corporate LDAP directory. Assuming that the user is authenticated successfully, the IdP returns a signed assertion which Jabber forwards to the collaboration service provider using HTTP POST. The collaboration service provider then validates the signed assertion and provides authorization to the Jabber client. For example, Jabber successfully registers to Unified CM.
Figure 21-35 Cisco Jabber with SAML SSO
In addition to forwarding a signed assertion to the Jabber client, the IdP stores a security context for the authenticated Jabber client. Should the client request access to other collaboration service providers, the IdP is able to provide subsequent signed assertions without requiring another exchange of credentials. In this way, SSO enables the Jabber user or client to access multiple collaboration services by entering their credentials once.
It is worth noting that the collaboration service provider never communicates directly with the IdP when authenticating the user.
For more information about SSO, refer to the Identity Management Architecture Overview, and the SAML SSO Deployment Guide for Cisco Unified Communications Applications available at
In addition to SSO user identification and authentication to on-premises collaboration applications and services, SAML SSO can also be enabled for user authentication over Expressway mobile and remote access connections. In these scenarios, an HTTPS reverse proxy is deployed in the DMZ of the enterprise to broker authentication for inbound remote access connections. The HTTPS reverse proxy communicates with the internal enterprise IdP and brokers the SAML request and authentication exchange between the remote client and the enterprise IdP. While the HTTPS reverse proxy in the DMZ can be any generic HTTPS reverse proxy, some IdP vendors offer an option to install an IdP instance in the DMZ to serve an IdP proxy role for brokering or proxying SSO SAML requests.
Interactions Between Cisco Jabber and Cisco Unified Mobility
The Cisco Jabber mobile clients can be integrated with Cisco Unified Mobility to leverage Cisco Single Number Reach, mid-call DTMF features, two-stage dialing, and single enterprise voicemail box mobile voicemail avoidance.
Integration with Unified Mobility requires the iPhone or Android dual-mode mobile phone number to be configured within Unified CM as a mobility identity associated with the Cisco Dual Mode for iPhone or Cisco Dual Mode for Android device. Once the mobile number is configured as a mobility identity within the system, Single Number Reach can be leveraged so that incoming calls to the user's enterprise number will be extended to the iPhone or Android dual-mode device through the mobile voice network as long as the iPhone or Android dual-mode device is not connected to the enterprise and not registered to Unified CM. In situations where the dual-mode device is connected to the enterprise, registered to Unified CM, and the client calling options are set so that inbound voice-over-IP calling is enabled ("Voice over IP" or "Autoselect" when the device is connected to a WLAN), an inbound call to the enterprise number will not be extended to the mobile voice network interface of the device. When the iPhone or Android dual-mode device is connected to the enterprise, only the WLAN or mobile data interface of the device will receive the inbound call. This prevents unnecessary consumption of enterprise PSTN gateway resources.
When handling enterprise calls through the cellular voice network, the iPhone or Android dual-mode device can invoke mid-call features by means of DTMF and perform desk phone pickup for any enterprise anchored call. The dual-mode device can also leverage Mobile Voice Access and Enterprise Feature Access two-stage dialing features when making outbound calls to route these calls through the enterprise and anchor them in the enterprise PSTN gateway.
In addition to configuring a mobility identity for the iPhone or Android dual-mode device, you can configure additional mobile phone numbers or off-system phone numbers as remote destinations and associate them to the Cisco Dual Mode for iPhone or Cisco Dual Mode for Android device within Unified CM. When associating the mobility identity and additional remote destinations to the dual-mode device, you do not have to configure a remote destination profile.
When mobile users are provisioned with multiple Cisco mobile clients across multiple mobile devices (for example, a user running Cisco Jabber for Android on their Android smartphone and Cisco Jabber for iPhone and iPad on their Apple iPad), associate the mobility identity with the dual-mode device (for example, Cisco Dual Mode for Android) rather than with the tablet device (Cisco Jabber for Tablet). Because the dual-mode device leverages functionality unique to the mobility identity, including dual-mode handoff and dial via office, the mobility identity should be associated to this device. Associate all other remote destinations to the same device as the mobility identity. Associating different remote destinations on different mobile client devices for the same user makes configurations more complex and troubleshooting issues more difficult.
For more information about the Cisco Unified Mobility feature set as well as design and deployment considerations, see Cisco Unified Mobility.
Interactions Between Cisco Jabber and Cisco Intelligent Proximity for Mobile Voice
The Intelligent Proximity for Mobile Voice feature is designed to enable hands-free audio for the cellular or mobile line of a dual-mode devices. For this reason, usually only calls on the cellular line of the Jabber client device are enabled for hands-free audio play out on an Intelligent Proximity-capable IP endpoint. In the case of voice or video over IP calls on Cisco Jabber, Intelligent Proximity for Mobile Voice is not invoked. The one exception to this is with the Cisco IP Phone 8851 and 8861 endpoints. Because these IP phones are audio-only, with Intelligent Proximity for Mobile Voice, audio for a Jabber IP-based call is streamed through the 8851 or 8861 phone while the video portion of this call remains on the Jabber client device. In the case of other hardware endpoints capable of Intelligent Proximity for Mobile Voice, audio for Jabber IP-based calls is not played by the IP endpoint.
The Cisco Spark mobile client is available for Android and Apple iOS mobile devices, including iPad and iPhone. Once the client application is downloaded from the appropriate application store (Apple Application Store or Google Play) and installed on the Apple iOS or Android device, users must enter their email address and activate their account with the resulting provisioning email. Once a user activates their account, the client connects to the Cisco Collaboration Cloud and the user can begin creating secure collaboration rooms with one or more people to communicate using encrypted instant messaging (IM). The user should access Cisco Spark at http://web.ciscospark.com/ using a web browser at least once in order to set a password for their account. Alternatively, the user can use the desktop Cisco Spark client available for download from http://download.ciscospark.com/. Failure to do this will require the user to activate their account via email each time they connect with the mobile client.
Cisco Spark for Android, iPad, and iPhone clients not only provide secure persistent IM collaboration rooms, but they also provide encrypted voice and video calling over IP and file sharing capabilities.
For proper Cisco Spark client operation, the mobile device must be able to reach the Internet by connecting to a wireless network (enterprise or public/private 802.11 WLAN or mobile provider data network).
For more information about the Cisco Spark mobile clients, additional feature details, and supported hardware and software versions, refer to the Cisco Spark documentation at
Cisco WebEx Meetings
The Cisco WebEx Meetings mobile client runs on specific Android, Apple iOS, BlackBerry, and Windows Phone mobile devices. This client enables mobile endpoints to participate in Cisco WebEx Meetings with a similar experience as with desktop browser-based Cisco WebEx Meetings. This client enables active participation in Cisco WebEx voice and video conferencing, including the ability to view participant lists and shared content.
For more information about Cisco WebEx mobile clients, refer to the product information at
Cisco Cloud Collaboration Services: SAML SSO for Cisco Spark and Cisco WebEx
Just as with on-premises enterprise and collaboration edge deployments described earlier, enterprise SSO can be used to facilitate secure logins to cloud collaboration services such as Cisco Spark and Cisco WebEx. With these types of deployments the enterprise IdP in combination with an HTTPS reverse proxy deployed in the enterprise DMZ leverage enterprise credentials to identify and authenticate user access to Cisco Spark and Cisco WebEx.
Cisco AnyConnect Mobile Client
The Cisco AnyConnect mobile client provides secure remote connectivity capabilities for Cisco Jabber mobile device clients, enabling connectivity over mobile data networks and non-enterprise WLANs. The Cisco AnyConnect mobile client can be downloaded from the Apple Application Store or Google Play (formerly Android Market). This client application provides SSL VPN connectivity for Apple iOS and Android mobile devices through the Cisco AnyConnect VPN solution available with the Cisco Adaptive Security Appliance (ASA) head-end.
When employing VPN network connectivity for connections over the mobile data network or public or private Wi-Fi hot spots, it is important to deploy a high-bandwidth secure VPN infrastructure that adheres to the enterprise's security requirements and policies. Careful planning is needed to ensure that the VPN infrastructure provides high bandwidth, reliable connections, and appropriate session or connection capacity based on the number of users and devices using this connectivity.
For more information on secure remote VPN connectivity using Cisco AnyConnect, refer to the Cisco AnyConnect Secure Mobile Client documentation available at
High Availability for Cisco Mobile Clients and Devices
Although mobile devices and in particular dual-mode phones by their nature are highly available with regard to network connectivity (when the WLAN network is unavailable, the mobile voice and data networks can be used for voice and data services), enterprise WLAN and IP telephony infrastructure high availability must still be considered.
First, the enterprise WLAN must be deployed in a manner that provides redundant WLAN access. For example, APs and other WLAN infrastructure components should be deployed so that the failure of a wireless AP does not impact network connectivity for the mobile device. Likewise, WLAN management and security infrastructure must be deployed in a highly redundant fashion so that mobile devices are always able to connect securely to the network. Controller-based wireless LAN infrastructures are recommended because they enable centralized configuration and management of enterprise APs, thus allowing the WLAN to be adjusted dynamically based on network activity and AP failures.
Next, remote secure connection solution components, including the Cisco ASA head-end VPN terminator and the Cisco Expressway-E and Expressway-C nodes, should be deployed in a highly redundant fashion so that loss of a Cisco ASA or a Cisco Expressway node does not impact or prevent secure mobile and remote access connectivity for the mobile client.
Next, Unified CM call processing and registration service high availability must be considered. Just as with other devices within the enterprise that leverage Unified CM for call processing services, mobile client devices must register with Unified CM. Given the redundant nature of the Unified CM cluster architecture, which provides primary and backup call processing and device registration services, mobile device registration as well as call routing are still available even in scenarios in which a Unified CM node fails.
Similar considerations apply to PSTN access. Just as with any IP telephony deployment, multiple PSTN gateways and call routing paths should be deployed to ensure highly available access to the PSTN. This is not unique to mobile client device deployments, but is an important consideration none the less.
In the case of the Cisco Collaboration Cloud, WebEx and Cisco Spark services are highly available due to the redundant component and resource design in the cloud data centers, including both compute and network access platforms. This resilient infrastructure design provides highly reliable access for Cisco mobile clients that rely on Cisco Collaboration Cloud services.
Capacity Planning for Cisco Mobile Clients and Devices
Capacity planning considerations for Cisco mobile clients and devices, including dual-mode phones, are the same as for other IP telephony endpoints or devices that rely on the IP telephony infrastructure and applications for registration, call processing, and PSTN access services.
When deploying Cisco mobile clients and devices with Unified CM, it is important to consider the registration load on Unified CM as well as the Unified Mobility limits. A single Unified CM cluster is capable of handling a maximum of 40,000 device configurations and registrations. When deploying mobile clients and devices, you must consider the per-cluster maximum device support, and you might have to deploy additional call processing clusters to handle the added load.
In addition, as previously mentioned, the maximum number of remote destinations and mobility identities within a single Unified CM cluster is 40,000. Because most dual-mode mobile client devices will likely be integrated with Unified Mobility to take advantage of features such as Single Number Reach, single enterprise voicemail box mobile voicemail avoidance, desk phone pickup, and two-stage dialing, the mobile phone number of each of these dual-mode mobile devices must be configured as a mobility identity within the Unified CM cluster. This is necessary to facilitate integration to Unified Mobility as well as to facilitate the Handoff Number method of hand-out. Therefore, when integrating these dual-mode devices with Unified Mobility, it is important to consider the overall remote destination and mobility identity capacity of the Unified CM cluster to ensure sufficient capacity exists. If additional users or devices are already integrated to Unified Mobility within the system, they can limit the amount of remaining remote destination and mobility identity capacity available for dual-mode devices.
Another scalability consideration for Cisco mobile clients is the Cisco Expressway mobile and remote access call and proxy registration capacity of the Expressway-C and Expressway-E nodes. Expressway-C and Expressway-E clusters support a maximum of 10,000 proxy registrations and a maximum of 2,000 video or 4,000 audio calls. When determining available capacity for Cisco mobile clients, remember to include other Expressway attached devices – for example, Jabber desktop clients and fixed endpoints such as Cisco TelePresence MX and SX Series devices, and Cisco desk phones such as the 7800 and 8800 Series devices – in the calculations. Likewise, registration load on Unified CM cluster nodes must also be considered for Cisco Mobile client devices connecting to the enterprise through Expressway mobile and remote access. See Cisco Expressway, for more details on Cisco Expressway mobile and remote access sizing.
Overall call processing capacity of the Unified CM system and PSTN gateway capacity must also be considered when deploying mobile client devices. Beyond handling the actual mobile device configuration and registration, these system must also have sufficient capacity to handle the added BHCA impact of these mobile devices and users. Likewise, it is critical to ensure sufficient PSTN gateway capacity is available to accommodate mobile devices. This is especially the case for dual-mode mobile devices that are integrated to Unified Mobility because the types of users that would have dual-mode devices are typically highly mobile. Highly mobile users typically generate more enterprise PSTN gateway load from mobility features such as Single Number Reach, where an incoming call to a mobile user's enterprise number generates one or more calls to the PSTN, or from two-stage dialing, where a user makes a call through the enterprise by leveraging the enterprise PSTN gateway.
Finally, just as with enterprise mobility deployments, 802.11 WLAN call capacity must be considered when deploying Cisco mobile clients and device. As previously mentioned, a maximum of 27 VoWLAN calls or a maximum of 8 VVoWLAN calls are possible per 802.11 channel cell. This assumes no Bluetooth when devices are deployed on the 2.4 GHz band, 24 Mbps or higher data rates for VoWLAN calls, and 720p video resolution with bit rates up to 1 Mbps for VVoWLAN calls. Actual call capacity could be lower depending on the RF environment, wireless endpoint type, and WLAN infrastructure. See Capacity Planning for Campus Enterprise Mobility, for more details regarding 802.11 WLAN call capacity.
The above considerations are certainly not all unique to mobile clients and devices. They apply to all situations in which devices and users are added to Unified CM, resulting in additional load to the overall system.
For more information on general system sizing, capacity planning, and deployment considerations, see the chapter on Collaboration Solution Sizing Guidance.
Design Considerations for Cisco Mobile Clients and Devices
Observe the following design recommendations when deploying Cisco mobile clients and devices:
- Dual-mode mobile devices must be capable of dual transfer mode (DTM) in order to be connected simultaneously to both the mobile voice and data network and the WLAN network so that the device is reachable and able to make and receive calls on both the cellular radio and WLAN interface of the device. In some cases, proper dual-mode client operation might not be possible if mobile voice and data networks do not support dual-connected devices.
- WLAN APs should be deployed with a minimum cell overlap of 20%. This overlap ensures that a mobile device can successfully roam from one AP to the next as the device moves around within a location, while still maintaining voice and data network connectivity.
- WLAN APs should be deployed with cell power level boundaries (or channel cell radius) of -67 dBm in order to minimize packet loss. Furthermore, the same-channel cell boundary separation should be approximately 19 dBm. A same-channel cell separation of 19 dBm is critical for ensuring that APs or clients do not cause co-channel interference to other devices associated to the same channel, which would likely result in poor voice and video quality.
- Whenever possible rely on the 5 GHz WLAN band (802.11a/n/ac) for connecting mobile clients and devices capable of generating voice and video traffic. 5 GHz WLANs provide better throughput and less interference for voice and video calls.
- The enterprise wired and wireless LAN should be deployed and configured to support the necessary end-to-end QoS classes of service, including priority queuing for voice media and dedicated video and signaling bandwidth, to ensure the quality of client application voice and video calls and the appropriate behavior of all features. While most clients mark traffic appropriately at Layer 3 based on Cisco QoS recommendations, appropriate Layer 2 WLAN UP marking is dependent on the client device and vendor implementation. For this reason, Layer 2 marking is not consistent across platforms and as such cannot be relied upon.
- Because mobile devices are similar to desktop computers and can generate a large variety of data and real-time traffic, these devices are typically considered untrusted. For this reason, the network should be configured to re-mark all traffic from these client devices based on port number and/or protocol. Likewise, rate limiting and policing on ingress to the network is recommended.
- Cisco recommends using only an enterprise-class voice and video optimized WLAN network for connecting mobile devices and clients. While most mobile client devices are capable of attaching to public or private WLAN access points or hot spots for connecting back to the enterprise through the Internet for call control and other collaboration services, Cisco cannot guarantee voice and video quality for these types of connections.
- When deploying Cisco collaboration mobile clients and devices on a Cisco Bring Your Own Device (BYOD) infrastructure, administrators should consider a network attachment method that does not require user intervention and which maximizes utilization of the IP telephony infrastructure. Further, for remote connectivity scenarios, all relevant ports must be opened in the corporate firewall in order for Cisco mobile clients and devices to be able to access collaboration services.
- If corporate policy dictates that the BYOD infrastructure must remotely wipe or factory-reset lost or stolen mobile devices, employees using personal mobile devices should be aware of the policy and should regularly back up personal data.
- The Unified Mobility Single Number Reach feature will not extend incoming calls to the dual-mode device's configured mobility identity if the dual-mode device is inside the enterprise and registered to Unified CM. This is by design in order to reduce utilization of enterprise PSTN resources. Because the dual-mode device registers to Unified CM, the system knows whether the device is reachable inside the enterprise; and if it is, there is no reason to extend the call to the PSTN in order to ring the dual-mode device's cellular voice radio. Only when the dual-mode device is unregistered will Single Number Reach extend incoming calls to the user's enterprise number out to the mobility identity number on the PSTN.
- When you deploy mobile devices, Cisco recommends normalizing required dialing strings so that users are able to maintain their dialing habits, whether the mobile device is connected to the enterprise or not. Because dialing on the mobile network is typically done using full E.164 (with or without a preceding '+') and mobile phone contacts are typically stored with full E.164 numbers, Cisco recommends configuring the enterprise dial plan to accommodate full E.164 or full E.164 with preceding '+' for mobile client devices. By configuring the enterprise dial plan in this manner, you can provide the best possible end-user dialing experience so that users do not have to be aware of whether the device is registered to Unified CM.
- Cisco recommends that dual-mode phone users rely exclusively on the mobile voice network for making emergency calls and determining device and user location. This is because mobile provider networks typically provide much more reliable location indication than WLAN networks. To ensure that dual-mode phones rely exclusively on the mobile voice network for emergency and location services, configure the Emergency Numbers field of the dual-mode devices within Unified CM with emergency numbers such 911, 999, and 112 in order to force these calls over the mobile voice network. Dual-mode phone users should be advised to make all emergency calls over the mobile voice network rather than the enterprise network. Although making emergency calls over corporate WLANs or mobile data networks is not recommended, mobile devices that do not have cellular voice radios are capable of making calls only through these data interfaces. Mobile devices that do not have cellular voice radios should not be relied upon for making emergency calls.
- When deploying Cisco Jabber on mobile devices, configure the WLAN network to accommodate the following deployment guidelines:
– Minimize roaming of Cisco Jabber mobile client devices at Layer 3 on the WLAN. Layer 3 roaming, where a device IP address changes, will result in longer roam times and dropped voice packets and could even result in dropped calls.
– Configure the same SSID across all APs utilized by the Cisco Jabber mobile client devices within the WLAN to ensure the fastest AP-to-AP roaming.
– Configure all enterprise WLAN APs to broadcast their SSIDs in order to prevent mid-call prompts to join other APs within the WLAN infrastructure, which could result in interrupted calls.
- Provide sufficient wireless voice and video call capacity on the enterprise wireless network for Cisco mobile clients and devices by deploying the appropriate number of wireless APs to handle the desired call capacity based on mobility-enabled user BHCA rates. Each 802.11g/n (2.4 GHz) or 802.11a/n/ac (5 GHz) channel cell can support a maximum of 27 simultaneous voice-only calls with 24 Mbps or higher data rates. Each 802.11g/n (2.4 GHz) or 802.11a/n/ac (5 GHz) channel cell can support a maximum of 8 simultaneous video calls assuming 720p video resolution at up to 1 Mbps bit rate. For 2.4 GHz WLAN deployments, Bluetooth must be disabled to achieve this capacity. Actual call capacity could be lower depending on the RF environment, wireless endpoint type, and WLAN infrastructure.
- When deploying Dial via Office Reverse (DVO-R), use of the User Control method of voicemail avoidance ensures that called users do not end up in the calling user's voicemail box. This method of voicemail avoidance requires the calling user to press a number on the mobile device key pad in order to connect the DVO-R call. Failure to press a key on the mobile device results in the DVO call being cleared.
- DVO-R calls using the alternate callback number are not anchored in the enterprise and therefore desk phone pickup and DTMF-based mid-call features may not be used on these calls. In addition, voicemail avoidance is not engaged for calls to alternate callback numbers.
- The following features and capabilities are not supported over Expressway mobile and remote access connections: WLAN to cellular dual-mode handoff, LDAP directory access, per-user or per-device access restrictions, and session persistence during network path changes. If any of these features are required, consider implementing a Cisco AnyConnect VPN solution for Jabber mobile clients.
- For deployments of Cisco Unified CM versions prior to 11.5 and/or Jabber versions prior to 11.5, basic directory integration (BDI) with LDAP is the recommended on-premises directory access method for Jabber mobile clients because UDS directory access methods reduce Unified CM node registration capacity by 50% in those earlier Unified CM versions. When BDI or CDI is used for on-premises directory access, UDS will be used by Jabber mobile clients only when they connect through Expressway mobile and remote access.
- When mobile users are provisioned with multiple Cisco mobile clients across multiple mobile devices, the mobility identity and any additional remote destinations should always be associated to the Cisco Jabber dual-mode device type.
- After initially downloading, installing, and activating the Cisco Spark account via the mobile device, the user should access Cisco Spark using a web browser or desktop client in order to create a password for their account. Once this is done, the user will be able to access Cisco Spark using any client (mobile, desktop, or web browser). Failure to set a password results in the user having to re-activate their account through email after sign-out each time.