Appendix A. Answers to the “Do I Know This Already?” Questions

Chapter 1

1. B, C. Both the CUBE and Expressway-C products exist at the collaboration edge and usually interface with public networks such as the Internet or the public switched telephone network (PSTN).

2. B. SIP trunks are leveraged by nearly every device within the Cisco collaboration ecosystem for call routing and supplementary features.

3. A, B. Endpoint registration and ad hoc conferencing are features offered natively by Unified CM. The other options listed here are best left to Expressway and CUBE.

4. C, D. SIP and H.323 are VoIP protocols for which CUBE can perform interworking. MGCP and SCCP are used to control analog and TDM endpoints on voice gateways, and WebRTC is not supported on CUBE.

5. A. A numbering plan area (NPA) code, often referenced simply as an area code, is a three-digit code that is usually mapped to a location such as a city.

6. D. +1 is the country code for the United States, and 4085267209 is the national number in this case.

7. D. Most users within the United States dial the steering code followed by 011, which is the country code, and the phone number.

Chapter 2

1. D. A back-to-back user agent (B2BUA) or session border controller (SBC) is a device that can act as a UAS to answer incoming requests and then perform call routing functions and reoriginate requests by operating as a UAC.

2. B, E. 5xx-level messages are server error final responses.

3. A, E. Of the headers mentioned, only Call-ID and Contact are required during a SIP INVITE request.

4. A, C. During an early offer call, the SIP INVITE and 200 OK messages carry the SDP message body to complete the offer/answer.

5. C, D. OPUS audio and H.264 video both utilize dynamic payload numbers (96 through 127) and must therefore leverage the a=rtpmap SDP attribute.

6. A, C. Both INVITE and UPDATE can be used in concert with SDP to perform mid-call session updates to modify SDP session parameters.

7. B. H.245 is used to negotiate the characteristics of a media session, such as media format, the method of DTMF relay, the media type (audio, video, fax, and so on), and the IP address/port pair for media.

8. B. TCP port 1720 is used for non-secure signaling with the H.225 protocol.

9. B. When fast start is utilized, H.245 signaling is tunneled within the H.225 signaling and completes before the call connects.

Chapter 3

1. A. 8 kHz equals 8000 Hz, which is the sampling rate for the G.711 audio codec.

2. C, D, and E. Dynamic RTP payload types are 96 through 127.

3. D. The Sequence Number header is a 16-bit field that facilitates built-in loss detection for RTP packets

4. C. The SSRC value for a given RTP stream might change when the RTP network address and port pair change. The other scenarios do not have any bearing on the SSRC value.

5. A. The recommended percentage of bandwidth allocation for RTCP is 5%, and active senders are allocated one-quarter of the total RTCP bandwidth.

6. A, B. Both of these SDP attributes are used to signal the RTCP port that will be used for the stream.

7. B. RTP-NTE packets are RTP packets carrying the specialized named telephony event (NTE) DTMF payload.

8. D, E. Both 99 and 101 fall within the dynamic payload type number range that is used by RTP-NTE DTMF.

9. A. The initial SIP INVITE message contains the first digit dialed by the user. Subsequent digits are sent through the KPML subscription using SIP NOTIFY messages.

10. C. OOB DTMF is signaled through the call setup signaling. For SIP and H.323, the signaling should be examined to determine if there are any issues with negotiation or transmission of digits.

Chapter 4

1. B. The X character represents the digits 0 through 9 as a wildcard in Unified CM.

2. B, C, and E. A is invalid because masks do not use a leading backslash before a plus sign. B is a valid mask. C is valid because * is a character, not a wildcard, so the digits in those positions will be replaced with the * character. D is invalid because bracket notation is not allowed in a mask. E is valid.

3. A, B, and E. The Use Calling Party’s External Phone Number Mask parameter appears in the configuration pages for each of these dial plan elements.

4. B, D. Top Down and Circular are the two options available for a route group. Line groups also have the Longest Idle option, but this is not available on a route group.

5. C, D, and E. Digit discard instructions apply to patterns with the @ sign where numbering plan–specific instructions can be applied or to any pattern with a dot (.) or trailing pound symbol (#).

6. B. Call queuing can only be configured on a hunt pilot.

7. D. Calling search spaces are ordered lists of partitions. Route patterns and directory numbers are placed into partitions, not into calling search spaces.

8. B. Patterns in the < None > partition can be reached by any calling search space, including the < None > calling search space.

9. A, C, and D. All these parameters are available on the configuration for a translation pattern but not on a route pattern.

10. C. The translation pattern transforms the original called party number to 2000. When a transformation pattern is applied, it uses the original called party number to find a match for the transformation and as the input into any configured transformations; hence 2XXX is matched, and the called party number is transformed to 2222.

11. C. The order of operations is digit discard instructions, transformation mask, and then prefix digits, in this order, so 456888 becomes 8888 after discarding PreDot, it then becomes 118888 after applying the mask, and finally it becomes 123118888 after prefixing the digits.

12. C. The first step in a global dial plan with TEHO should be to globalize the calling and called party numbers so they can be routed in globalized form and then localized, depending on the egress gateway or trunk.

13. A, C. Only local directory URIs are placed into partitions. Learned URIs are not placed in a partition. SIP route patterns are also placed into a partition when configured. The CFQDN and OTLD parameters only apply to numeric URIs and are not applicable to non-numeric URIs.

14. B, C. A cluster can be either a hub or a spoke if it is in an ILS network. It can also be configured to be a Stand Alone Cluster, but in that case, it is not part of an ILS network.

15. C, D, and E. TranslatorX, RTMT, and CSA all have the ability to display a SIP ladder diagram for calls.

16. E. Options A, C, and D are possible outcomes, but the one line does not tell you which one is correct. This line just indicates that there are potential matches if the user continues to dial more digits but does not tell you if there has been a match. You must look for a digit analysis result to know if a call will be routed or not.

Chapter 5

1. B, D, and E. CUC, IM&P, and CUBE can integrate with Unified CM by utilizing a SIP trunk.

2. D. The inbound device is the SIP trunk that assumes the role of the calling device.

3. C. You define a port by using the Inbound Incoming Port option on the SIP trunk security profile applied the SIP trunk.

4. C. SIP OPTIONS ping can be used to monitor the reachability status of a remote device.

5. C. The behavior during an MTP allocation error depends on the Unified CM Fail Call Over SIP Trunk if MTP Allocation Fails service parameter. By default, the call proceeds rather than fails.

6. C. The Run On All Active Unified CM Nodes checkbox allows all Unified CM nodes, with the CallManager service enabled, to monitor and process calls with the SIP trunk, thus adding more redundancy.

7. B, D. Both of these methods allow you to gather information about the TCP handshake failure.

8. B. Unified CM writes information about SIP trunk operation to the CallManager SDL traces.

Chapter 6

1. A, B, C. The IOS-based media resources include MTPs, transcoders, and conference bridges.

2. B, C, D. Media resources either register using SCCP or, for some conference resources, a combination of SIP and HTTPS.

3. A, E. The IP Voice Media Streaming App service supports the G.711 and Wideband 256k codecs for audio conferencing.

4. B. The CMS certificate must be imported into the CallManager-Trust trust store.

5. C, D, E. Unified CM supports ad hoc, Meet-Me, and Conference Now as native conference types.

6. B. Most of the configuration for the Conference Now feature is done on the end-user configuration page.

7. B, D. The two hold source types are user hold and network hold.

8. D. You can configure up to 500 audio sources besides the fixed audio source.

9. C. A media resource group list configured on a device overrides the configuration from the device pool.

10. C. The Call Park Reversion timer is set to 60 seconds by default.

11. B. To retrieve a call parked on a directed call park number, the user must first dial the retrieval prefix and then dial the park number or use a directed park BLF button (not a standard speed dial button).

12. B, D, E. Unified CM supports pickup, group pickup, other pickup, and directed call pickup.

13. B. An audio codec preference list allows an administrator to reorder the list of codecs advertised by Unified CM, but the codecs cannot be removed from the list. There are service parameters that control removing some codecs.

14. A, B, E. Regions allow administrators to define bandwidth limits for audio, video, and immersive video calls.

15. B, C, D. ELCAC requires configuration of locations, links, and weights. LBM groups are recommended but not required, and the default region is all that is needed, although adding additional regions to match the locations being configured is highly recommended.

16. B, C, D. The Hub_None, Phantom, and Shadow locations are all preconfigured on Unified CM.

Chapter 7

1. B. Single Number Reach is a Unified Mobility solution provided by Unified CM that allows calls to a desk phone to ring a remote destination, such as a mobile phone, simultaneously.

2. C. Unified CM invokes Intelligent Session Control to avoid sending a call to the remote destination, where possible, and instead ring the desk phone when a call originates from an on-premises device.

3. B, D. A common misconception is that the calling search space is used to perform call routing operations when invoking the Single Number Reach feature. However, the rerouting calling search space is the correct CSS for this scenario. Another common issue is not checking the line association checkbox on the remote destination configuration page.

4. A. Unified CM performs digit analysis in tandem with digit analysis for the endpoint. The call to the Single Number Reach endpoint is then made when the Delay Before Ringing timer expires.

5. C. Move to Mobile is a Single Number Reach configuration that allows desk phone users to click a softkey to transition a call from the local desk phone to a remote destination seamlessly.

6. C. The Move to Mobile feature is invoked through the use of the Mobility softkey, which must be added to an IP phone because it is not a default softkey.

7. B. Simply missing the Enable Move to Mobile checkbox means the Move to Mobile feature does not work.

8. D. Extension Mobility is a key feature of Unified Mobility. The ability to log in to an IP phone and have settings applied dynamically is a great feature for any mobile worker.

9. A. An IP phone’s configuration page has a dynamic portion that shows information about the currently logged in Extension Mobility user.

10. A. Failing to select the Enable Mobility checkbox on the end-user page causes problems with Extension Mobility login.

11. C. This is a very common real-world scenario that many have encountered. The absence of an Extension Mobility subscription on a device profile causes some issues with logging out.

12. B. Device Mobility allows an end user to move his entire phone from one physical location to another.

13. D. To configure Device Mobility, you need to understand when roaming sensitive settings or Device Mobility–related information settings are applied to a phone.

14. B. The main part of this output, where the Unified CM trace logs states “added mobile device to Device Mobility Route Table,” indicates that the phone is roaming and is now using Device Mobility settings.

Chapter 8

1. B. CUBE always has an inbound call leg and an outbound call leg when a call traverses the SBC.

2. C, D. A call flow diagram shows Layer 7 application protocols and sometimes Layer 4 transport information. Layers 1, 2, and 3 are often included in topology diagrams but are not often included in call flow diagrams.

3. B. CUBE filters inbound dial peers based on a VRF instance on the ingress interface.

4. D. Dial peer–level configurations always have precedence over all other configurations when multiple variants exist.

5. B. CUBE always evaluates the topmost Via header in a SIP message before evaluating other URI matches and even before calling and called number match criteria commands.

6. B, C. For a dial peer to be eligible for outbound dial peer matching, both an outbound matching command and next-hop session information must be defined.

7. A, B, D. dial-peer hunt 0 defines an outbound dial peer match that uses longest match in phone number, explicit preference, and random selection.

8. B, E. e164-pattern-maps can be configured to summarize destination-pattern and incoming called-number statements, which leads to fewer dial peers in the configuration.

9. B, E. show call active voice brief can be used to view information about active calls such as total time of the call, send/received media packets, dial peer used on each call leg, and many other identifiers. show voip rtp connections can be leveraged to view local and remote media information that was negotiated by the signaling.

10. C, D. Packets do not naturally egress through virtual interfaces such as loopback and VLAN SVIs. This means in order to source packets from these types of interfaces, application signaling and media binding must be defined to spoof the sources of the packets.

11. B, C. The numeric calling number and numerical called number can be modified with translation profiles.

12. B. Outbound SIP profiles are used to change egress SIP messaging on a given call leg.

Chapter 9

1. C. You can use early-offer forced to force CUBE to send an early offer if a delayed offer is received.

2. B, D. The inclusion of SDP in a 18X message generally indicates that ringback will be in-band, as per the SDP attributes.

3. B, C, D. SIP devices use sendonly, inactive, and zero out IP (0.0.0.0) to remove media and signal a call hold.

4. D. The Refer-To header is used for dial peer lookups when CUBE has been configured to consume REFER requests.

5. C. Consult, or attended, transfers allow the transferor to connect a call with the transfer target that can be used to discuss the transfer before the transfer is completed.

6. C. A PRACK is required to confirm the offer/answer before the call is connected (answered), at which point an UPDATE with SDP can be used to change media parameters.

7. C. CUBE performs a silent discard and drops the packet.

8. A, B. Only IPv4 and IPv6 addresses are explicitly configurable using the IP trusted list feature.

9. B. G729r8 is the default audio codec for VOIP dial peers.

10. B. audio forced disables video and application SDP on CUBE.

11. A. To enable asymmetric payload support for DTMF, audio codes, and video codecs, you use the command asymmetric payload dtmf.

Chapter 10

1. C. Unified SRST provided a call agent to remote branch IP phones for local registration in the event of a WAN failure or outage that deems the remote Unified CM unreachable.

2. A, C, D. To register SIP IP phones with Unified CME, you need to define the maximum number of extensions (max-dn), the maximum number of phones (max-pool), and the source IP address for voice register global (source-address).

3.

4. E. With auto-registration, the SEPAAAABBBBCCCC.cnf.xml file does not get generated until a later step; as a result, the file XMLDefault.cnf.xml is handed out instead.

5. D. Virtual dial peer information (along with many other outputs about the IP phone) is detailed in the output of the show voice register pool command.

6. B. Without the command allow-connections sip to sip, the SIP application rejects SIP-to-SIP calls.

7. D. Parallel hunt groups, also known as call blast hunt groups, ring all members at the same time.

8. C. Unified CME sends an out-of-dialog SIP REFER to the IP phone with a multipart/mixed message body that contains information about the multicast paging IP address and port the phone should join.

9. B, D. To park the call in a basic park slot, simply click the Park softkey and let Unified CME pick an available park slot. To retrieve this call, simply dial the park slot directly.

10. A, B. Unified E-SRST supports for both shared lines and voice hunt groups when in Unified SRST mode.

11. B. Connection Monitor Duration is a timer that specifies “a common link qualifying time period for all the devices using a specific Unified SRST reference.”

12. B, C. The default mode for voice register global is Unified SRST, so only the max-pool and max-dn commands need to be defined.