Appendix A. Answers to Chapter-Ending Review Questions
Chapter 1
1 What are three tasks that a voice gateway performs?
Answer: Voice gateways perform some or all of the following tasks:
• Interfacing with the IP network and the PSTN or PBX
• Supporting IP call control protocols, in addition to TDM call control protocols
• Performing call setup and teardown for calls between the VoIP and PSTN networks by terminating and reoriginating the call media and signaling
• Providing supplementary services, such as call hold and transfer
• Relaying DTMF tones
• Supporting analog fax and modems over the IP network
2 What are two benefits of adding an IPIPGW between networks?
Answer: Placing an IPIPGW between networks gives added network privacy and additional security.
3 What are four tasks that an H.323 gatekeeper performs?
Answer: An H.323 gatekeeper can perform the following functions:
• Address resolution
• Call admission control
• Bandwidth control
• Zone management
• Security
• Call management
• Routing of call control signaling
4 Which call control protocol is an IETF standard that requires a call agent to function?
Answer: MGCP is a call control protocol that is an IETF standard and required a call agent to function.
5 Which call control protocol is an IETF standard, uses a distributed call-control model, and is able to control multiple types of media connections?
Answer: SIP is an IETF standard that uses a distributed call-control model and can control multiple types of media connections.
6 Which call control protocol is an ITU-T standard and uses a distributed call-control model?
Answer: H.323 is an ITU-T standard that uses a distributed call-control model.
7 What are three typical CallManager deployment scenarios?
Answer: Three typical CallManager deployment scenarios are as follows:
• Single site deployment
• Multisite deployment with centralized call control
• Multisite deployment with distributed call control
Chapter 2
1 What role does the call agent play when using an MGCP gateway?
Answer: A call agent is essential when you are using an MGCP gateway. It knows the dial plan, so it tells the gateway where to route calls. It controls the setup and teardown of connections to endpoints on the gateway, and thus the setup and teardown of calls.
2 List the control messages that MGCP uses.
Answer: NotificationRequest—RQNT
Notify—NTFY
CreateConnection—CRCX
ModifyConnection—MDCX
DeleteConnection—DLCX
AuditEndpoint—AUEP
AuditConnection—AUCX
RestartInProgress—RSIP
3 How does MGCP Backhaul function?
Answer: The gateway terminates ISDN Q.921 messages but places the Q.931 messages in a TCP packet and sends them to the CallManager for processing.
4 What three commands do you need to enable MGCP on a router and identify Cisco CallManager as its call agent?
mgcp
mgcp call-agent [ip-address|hostname] [port] 5ccm-manager mgcp
5 Why is there a need for DTMF relay in a VoIP network, regardless of the gateway protocol used?
Answer: When you are using codecs that compress voice, DTMF tones might become distorted or part of the signal might be lost. Then the receiver would not be able to respond correctly to the numbers that were sent.
Chapter 3
1 List at least four benefits of using H.323 as a gateway protocol.
• Caller ID from FXO and T1 CAS
• Use of a fractional PRI
• Interoperability with applications and devices from other vendors
• No version dependence between a gateway and CallManager
• Granular call control
• Tcl and VXML applications can be used
• Support for legacy systems
• Support for more types of TDM interfaces and signaling than MGCP
• Multimedia support
• NFAS support
• Use of a gatekeeper for call control and address resolution
• PRI call preservation
2 When you are configuring an H.323 gateway on a CallManager, what information do you enter for Device Name?
Answer: Enter the IP address of the gateway. If CallManager can reach the gateway with more than one IP address, use the h323-gateway voip bind srcaddr ip-address command to select which address to use.
3 Name four types of DTMF Relay that H.323 uses.
Answer: H.245 uses Cisco-RTP, RTP-NTE, H.245 Alphanumeric, and H.245 Signal.
4 What commands must you enter for a Cisco gateway to use H.323 Fast Start?
Answer: Cisco gateways assume that H.323 Fast Start is used. No commands are necessary to configure it.
5 Briefly describe toll bypass.
Answer: With toll bypass, calls between company locations are sent over the IP network as VoIP when bandwidth is available, and they are routed over the PSTN when no bandwidth is available.
6 If you configure conflicting commands globally under the voice service voip configuration mode, and under the dial peer using voice class, which commands will the router use?
Answer: The router will use the most explicating applied commands, which in this case are the commands under the dial peer. It will use global commands only if more explicit ones are not configured.
7 How do you prevent active calls from being terminated when the CallManager becomes unreachable?
Answer: Turn off H.225 keepalives between the gateway and CallManager. The commands to do this are as follows:
GW1(config)#voice service voip
GW1(conf-voi-serv)#h323
GW1(conf-serv-h323)#no h225 timeout keepalive
Chapter 4
1 What do the acronyms UAC and UAS stand for? Define the difference between the two entities.
Answer: UAC stands for user agent client, and UAS stands for user agent server. The UAC originates the SIP session, and the UAS is the other endpoint of the session. It responds to INVITES and other messages sent from the UAC.
2 Name five types of SIP servers, and describe what they do.
Answer: The five types of SIP servers are proxy, registrar, location, redirect, and presence. Following is what each one does:
Proxy server—Receives SIP requests from a UAC and forwards them to the next-hop device, such as a SIP server or the UAS.
Registrar server—Registers UAs and provides location information upon request.
Location server—Maintains the location database for registered UAs.
Redirect server—Informs a UA about alternate locations for a called party that has moved.
Presence server—Gathers presence information and subscriptions, and sends notifications of events.
3 Name the five types of SIP methods that Cisco routers can both generate and respond to. What is the purpose of each one?
Answer: The five types of SIP methods that Cisco routers can generate and respond to are REGISTER, INVITE, ACK, CANCEL, and BYE. Following is what each one does:
REGISTER—Registers a UA address with a SIP server.
INVITE—Requests that a UA participate in a session.
ACK—Sent by the UAC when it has received the final response to an INVITE.
CANCEL—Ends a pending session, not one that has already been accepted.
BYE—Ends a call or session.
4 What command configures a dial peer to use SIP in its communication with its VoIP peer?
Answer: The session protocol sipv2 command under dial-peer configuration mode configures a dial peer to use SIP in its communication with its VoIP peer.
5 How does CallManager 4.x interact with a SIP gateway?
Answer: CallManager 4.x can register a SIP trunk to the gateway. The trunk is referred to as a signaling interface. It can also trunk to a SIP server, a CME gateway, or another CallManager cluster.
6 What additional SIP capabilities does CallManager 5.x add?
Answer: CallManager 5.x can register SIP phones and can function as a B2BUA.
7 Which Layer 4 protocol does SIP use by default, and what is the default port?
Answer: SIP uses UDP by default, but you can configure it to use TCP. The default port number is 5060.
8 What is the function of the SDP in SIP call setup?
Answer: SDP is used to exchange information about endpoint capabilities and negotiate call features, such as in an INVITE message. The SDP part of a SIP message contains information, such as the SDP version, the calling party organization, the IP address, expected media, and media attributes.
Chapter 5
1 Which port type should you use to connect to a two-wire analog service connecting to the PSTN for both inbound and outbound calling?
Answer: A two-wire analog service from the PSTN is considered an FXS port. You should connect this to an FXO port in a gateway. You could connect an FXS-DID port to the PSTN, but this configuration would support outbound dialing only.
2 What types of signaling are required on a voice circuit?
Answer: Supervisory signaling, address signaling, and informational signaling are required on a voice circuit. Supervisory signaling is used to indicated on-hook and off-hook status. Address signaling is used to convey destination information. Finally, informational signaling is used to provide user feedback, such as dial tone and ringing.
3 What is the difference between CAS and CCS on an E1 circuit?
Answer: Although both CAS and CCS use timeslot 16 to carry signaling, CAS has specific bits within a multiframe that are dedicated, or associated to each channel. With CCS, messages are used to carry signaling information.
4 What is the difference between SF and ESF?
Answer: An SF consists of 12 T1 frames. The framing bit is used for synchronization. Robbed-bit signaling can provide two signaling bits per channel (A and B).
An ESF consists of 24 T1 frames. In addition to synchronization, the framing bits can be used to provide data link information and CRC error correction. Robbed-bit signaling can provide 4 signaling bits per channel (A, B, C, and D).
5 What two types of echo are possible on a voice circuit?
Answer: The two possible types of echo on a voice circuit are acoustic echo and hybrid echo.
Acoustic echo is caused by poor acoustic isolation between the speaker and microphone.
Hybrid echo is caused by an impedance mismatch in a 2-wire to 4-wire hybrid.
6 Which signaling type supports ANI on T1 CAS circuits?
Answer: E&M-FGD supports reception of ANI. FGD-EANA supports transmission of ANI.
7 Which line signaling method should you use on an E1 R2 satellite link?
Answer: You should use R2-Pulse on an E1 R2 satellite link.
8 What component of an ISDN message is used to carry information about the call?
Answer: Call-specific information is carried in the Information Element of an ISDN message.
Chapter 6
1 What is a major drawback of using analog FXO trunks for inbound calls?
Answer: The drawback is that no DNIS information is provided, so calls cannot be directly routed to their destination.
2 Which command must you include when configuring a CAMA type 4 (KP-NPD-NXX-XXXX-ST) trunk?
Answer: You must use the ani mapping command to associate three-digit NPA numbers to a single-digit NPD.
3 You issue a show controller command on a digital trunk and see accumulated slip seconds in every interval. What could be the cause?
Answer: Incorrect clocking configuration on the physical port can cause slips. For a PSTN connection, line clocking is typically used.
4 When you are trying to set up Q.931 backhaul on an ISDN PRI using CallManager MGCP, the isdn bind-l3 ccm-manager command is not recognized. What could cause this?
Answer: The cause could be that you omitted the service mgcp keywords from the PRI group configuration on the controller for the physical interface.
5 What must you do prior to configuring a PRI group on an ISDN PRI circuit?
Answer: You must configure an ISDN switch type globally on the Cisco voice gateway prior to configuring a PRI group on an ISDN PRI circuit.
6 Which signaling type is necessary if you want to receive ANI information on a T1 CAS trunk?
Answer: E&M FGD is necessary if you want to receive ANI information on a T1 CAS trunk.
7 What are the two components of R2 signaling?
Answer: Line signaling (supervisory control signals) and interregister signaling (call setup control) are the two components of R2 signaling.
8 If it is necessary to use the cas custom command to modify the signaling on an E1 R2 trunk, what is the recommended first step?
Answer: Begin by configuring the country country use-defaults command under cas custom.
Chapter 7
1 How will caller ID work on an FXO to FXS connection between a PBX and a Cisco voice gateway?
Answer: Caller ID will work only in one direction, from FXS to FXO.
2 Which E&M trunk type is most commonly in use?
Answer: Type I is used most often in North America, and Type V is used most often throughout the rest of the world.
3 A four-wire E&M Type V trunk uses how many physical wires?
Answer: A four-wire E&M Type V trunk uses six wires.
4 Which command confirms the status of clocking for the TDM bus of a voice gateway?
Answer: The show network-clocks command lists available clock sources by priority.
5 How can you control the master/slave Layer 2 function of an ISDN PRI trunk?
Answer: You can control the master/slave Layer 2 function of an ISDN PRI trunk by using the isdn protocol-emulate user | network command.
6 What type of signaling is configured on DS0s that are used for T-CCS?
Answer: External signaling is configured using the type ext-sig parameter of the ds0-group command.
7 How do you configure the D-channel for an MGCP-controlled ISDN PRI trunk?
Answer: Configure Layer 3 backhaul to CallManager using the isdn bind-l3 ccm-manager command.
8 When you are using an MGCP-controlled PRI trunk, how can you verify the status of the ISDN Q.931 backhaul?
Answer: The show ccm-manager command shows the status of the backhaul channel.
Chapter 8
1 Voice is sensitive to delay, jitter, and packet drops. What are the recommended maximum values for each of these?
Answer: High-quality voice and interactive video have the following network requirements:
• A maximum of 150 ms of one-way delay
• A maximum of 30 ms jitter
• A maximum of 1 percent packet loss
2 When you use the Modular QoS CLI, or MQC, what steps are involved in setting a bandwidth limit for voice traffic that is sent out an interface?
Step 1. Classify the traffic using a class map.
Step 2. Create a policy map, and associate the class map with the policy map.
Step 3. Assign bandwidth to that class within the policy map.
Step 4. Assign the policy to an interface with the service-policy outbound policy-map-name command.
3 Data packets can be large compared to voice packets. Why is this a problem across a WAN link, and how can you remedy it?
Answer: Although voice might be placed in a priority queue, a voice packet can be delayed by a larger data packet if it arrives at the interface after the data packet has begun being serialized. To maintain the delay budget for voice, you should serialize all packets in 10 ms or less. This is not a problem on links of T1 speed or better, because you can serialize a 1500-byte packet within 10 ms. To remedy this, you can use LFI on slower-speed links.
4 What is the difference between fax/modem relay and passthrough?
Answer: In relay, the fax or modem analog data is demodulated by the sending gateway, packetized, and sent over the IP network. The receiving gateway remodulates it and forwards it as analog data to the fax or modem.
In passthrough, fax and modem calls are treated as any other analog voice call, with the data carried in-band in RTP packets to the remote fax or modem.
5 What are the two types of fax relay that Cisco routers use, and which is the default type?
Answer: Cisco routers use Cisco proprietary fax relay and the ITU-T standardized T.38 fax relay. Cisco fax relay is the default.
6 In which configuration mode are fax/modem commands given on MGCP gateways? How about H.323 and SIP gateways?
Answer: Fax/modem relay and passthrough commands are given at global configuration mode on MGCP gateways. On H.323 and SIP gateways, they can be given at either dial peer or voice service configuration mode.
7 How does SRTP protect voice media traffic?
Answer: SRTP encrypts the RTP voice media payload, but not the RTP header, using AES encryption. It authenticates the RTP header and payload contents by computing a one-way HMAC-SHA1 hash and placing the results in an authentication tag at the end of the packet. The receiver runs the same computation and compares its result to the contents of the authentication tag. If the contents do not match, the receiver drops the packet. SRTP also includes a replay protection process to avoid DoS attacks.
8 When using encrypted voice within a LAN, why is it a good idea to also encrypt traffic between the voice gateway and Cisco CallManager?
Answer: Voice media and signaling are not the only types of voice traffic that traverse the WAN. MGCP gateways communicate with Cisco CallManagers. IP phones download TFTP files and Dynamic Host Configuration Protocol (DHCP) information, if those servers are centrally located. DTMF tones and encryption keys might be exchanged. If you are encrypting the IP phone traffic, it makes sense to encrypt the other voice traffic also, unless your network is secure and trusted.
9 How is firewall function affected if an IPsec tunnel from a remote gateway terminates on the Cisco CallManager, rather than another device?
Answer: A firewall cannot thoroughly inspect IPsec traffic coming from the WAN and going through the firewall into the LAN. All the firewall sees is the IPsec header. It would have to be able to decrypt the packet to see the original headers and the payload information. Thus, terminating an IPSec connection on the CallManager prevents the firewall from inspecting that communication, but it also secures that traffic while it traverses the network to reach the CallManager.
Chapter 9
1 What determines whether a gateway does digit-by-digit matching?
Answer: A gateway uses digit-by-digit matching if the incoming dial peer is not configured for direct-inward-dial. If the incoming dial peer is configured for direct-inward-dial, the entire digit string is matched. The exception is if the incoming call is ISDN with overlapping receiving.
2 What is the default order of operation for matching outbound dial peers?
Answer: Most specific match is the first selection. If two equivalent matches exist, the peer with the highest configured preference is selected. If preferences are equal, the selection is random.
3 What is necessary for a dial peer to be considered operational?
Answer: You must configure the dial peer with an answer address or an incoming called number or both a destination pattern and a target.
4 Which factors must you balance when designing a dial plan?
Answer: You must balance ease of use for end users with administrative overhead and scalability.
5 What end-user issue can be caused by an overlapping dial plan?
Answer: An overlapping dial plan requires the interdigit timer to expire before the appropriate route pattern can be selected. This results in a long post dial delay. You can minimize this issue by reducing the interdigit timer or by training users to use the # to indicate they have entered all digits.
6 What is the default dial peer?
Answer: The default dial peer is a system dial peer that is used to match inbound calls if the call does not match a configured dial peer.
7 Which number is used to match the destination pattern on a dial peer?
Answer: For inbound dial peers, the calling number (ANI) is used to match the destination pattern. For outbound dial peers, the called number (DNIS) is used to match the destination pattern.
8 How are dial peers configured to accommodate an overlapping dial plan?
Answer: For the dial peers that have overlapping destination patterns, you need to add a T to the end of the digit string. This signifies that the digit matching should not occur until the interdigit timer expires.
Chapter 10
Answer: Digit manipulation encompasses adding, subtracting, and changing telephone numbers.
2 Of the digit manipulation techniques digit stripping, digit forwarding, digit prefixing, number expansion, and CLID, which are executed after the outbound dial peer is matched but before the numbers are transmitted?
Answer: All but number expansion are executed after the outbound dial peer is matched but before the numbers are transmitted.
3 By default, POTS dial peers strip any outbound digits that explicitly match their destination pattern. What are two simple ways to prevent the router from stripping all the digits?
Answer: Use either the no digit-strip command or the forward-digit number command under dial-peer configuration mode.
4 When is a number expansion executed, and how can you test its action?
Answer: A number expansion is a global policy that is executed before any outbound dial peer is matched. The command show dialplan number number can test its action.
5 Given the following voice translation rule, how would a dialed string of 913012345678 be translated?
/^(91)301(.......)/ /12/
Answer: It would translate to 12345678. The 91 is translated to 1, and then the contents of the second set (specified with the 2) are inserted. Because 301 is neither translated nor inserted, it is ignored.
6 Given the following voice translation rule, how would a dialed string of 913012345678 be translated?
/^(91)301(.......)/ /17002/
Answer: It would translate to 917002345678. The contents of the first set are inserted, the 301 is translated to 700, and then the contents of the second set are inserted.
Chapter 11
Answer: Three types of CAC include local CAC, measurement-based CAC, and resource-based CAC.
2 What CAC mechanism would you use to guarantee enough bandwidth for the duration of a call?
Answer: You could use RSVP, which reserves bandwidth on a call-by-call basis and rejects calls when any router in the path is unable to provide sufficient resources.
3 How are dial peers configured to be part of the same hunt group?
Answer: You can use a destination pattern that points the dial peers to the same phone numbers, and designate a preference value to control their order of use.
4 What is the difference between LVBO and AVBO?
Answer: LVBO monitors router interfaces and makes the busyout decision based on the state of the interface. AVBO sends a probe to measure network congestion and make its busyout decision based on the results of that probe.
5 Which CAC mechanism, other than AVBO, uses probes in making its call admission decision?
Answer: PSTN fallback also uses probes to make its call admission decision.
6 What is TEHO used for, and what are two issues with its use?
Answer: TEHO is used to minimize long-distance toll charges. Issues with its use include dial plan complexity and regulatory restrictions.
7 What are some differences between gateway-controlled RSVP and CallManager-controlled RSVP?
Answer: You can use gateway-controlled RSVP only with gateway protocols that have dial peers, because it requires some configuration under the dial peers. The gateway makes the CAC decisions based on its RSVP policy configuration.
You can use CallManager-controlled RSVP with all gateway protocols because it does not require dial peer configuration. RSVP policy configuration is done on CallManager, and CAC decisions are controlled by CallManager. CallManager-controlled RSVP uses SCCP to communicate with a media resource called an RSVP agent on the gateway.
8 Given the following dial peers, which would the gateway use first? Second? Third?
dial-peer voice 2200 voip
preference 1
destination-pattern 2200
session target ipv4:10.20.25.1
dial-peer voice 2201 voip
destination-pattern 2200
session target ipv4:10.20.26.2
dial-peer voice 2202 voip
preference 4
destination-pattern 2200
session target ipv4:10.20.27.3
Answer: Dial peer 2201 would be the most preferred. It has the default preference value of 0, which is the highest priority. (Lower preference values have higher priority.) Dial peer 2200 would be used second because its preference value is 1, and dial peer 2002 would be least preferred because it has the highest preference value.
Chapter 12
1 What conditions must exist for COR to restrict a call?
Answer: Both the inbound and outbound dial peers must have a COR list defined, and the outgoing COR list must be a subset of the incoming COR list.
2 An outgoing COR list is analogous to which Cisco CallManager Class of Control setting?
3 An incoming COR list is analogous to which Cisco CallManager Class of Control setting?
4 How many COR lists can you assign in SRST?
Chapter 13
1 Which two steps are required to enable MGCP Gateway Fallback?
Answer: Enable MGCP Gateway Fallback using the ccm-manager fallback-mgcp command, and set the alternate application to default.
2 Which device initiates SRST?
Answer: The SRST process is initiated by the IP phone sending a registration request to the SRST gateway.
3 Which gateway protocol can provide call preservation when using ISDN to the PSTN?
Answer: H.323 can provide call preservation when using ISDN to the PSTN.
4 Which command supports both extension addressing and E.164 addressing?
Answer: dialplan-pattern supports both extension addressing and E.164 addressing.
5 Which two SIP functions does a gateway perform to enable SIP SRST?
Answer: A gateways performs SIP Registrar Server and B2BUA to enable SIP SRST. Version 3.0 used Redirect Server instead of B2BUA.
6 Which certificate server can a Secure SRST router autoenroll with?
Answer: A Secure SRST router can autoenroll with Cisco IOS certificate server.
7 Which gateway protocol should you use for a remote site with an E1 PRI to provide the best call preservation?
Answer: You should use H.323 for a remote site with an E1 PRI to provide the best call preservation.
8 What is required to support integration to a centralized voice-mail system over a PRI?
Answer: The service provider must pass RDNIS. It might also be possible to support RDNIS using the forward-digits extra inband command on the dial peer if the router is running Cisco IOS 12.3(11)T or later.
Chapter 14
1 What is the purpose of a DSPFarm Profile?
Answer: The DSPFarm profile allows the DSPs of a gateway to be registered to multiple CallManager groups.
2 When is a DSP needed for an MTP?
Answer: You need a DSP when the packetization period for the voice streams is not the same. If the packetization period is the same, you can use a software-based MTP.
3 What is the default codec complexity for a C5510 DSP?
Answer: Flex complexity is the default codec complexity.
4 What is the major difference in the way C549 and C5510 DSPs handle voice termination?
Answer: When you are using a C549 DSP, voice termination is assigned statically. Each voice port is permanently assigned to a DSP when the voice port is created. When you are using a C5510 DSP, voice termination is dynamically assigned. Sequential calls on the same voice port might be handled by different DSPs.
5 You need to terminate 14 G.711 calls, provide transcoding for 6 G.729a to G.711 calls, and support 5 G.711 conference sessions with up to eight participants in each conference. What is the minimum number of DSPs required?
Answer: You need three C5510 DSPs. C549 DSPs support only six participants per conference. One DSP supports the voice termination for 14 calls. One DSP supports the six required medium-complexity transcoding sessions. One DSP supports up to eight G.711 conference calls.
6 You need to support one mixed-mode conference with up to eight participants. How should you configure the maximum sessions under the conference profile?
Answer: Because a DSP that is configured for conferencing cannot support another function, you should configure the maximum sessions to 2, which is the maximum number of conferences that a C5510 DSP can support. Setting the maximum sessions to 1 eliminates the possibility of using the DSP to its full capability.
7 Describe two benefits of DSP sharing.
• It reduces the possibility of oversubscribing DSP resources.
• It makes it easier to add DSP resources. DSPs can be added to an NM instead of having to dismantle the router to access the main board.
8 Your ISR 2811 has an NM-HDV with five PVDM-12s and two PVDM2-48s installed on the main board. The NM-HDV is configured for medium complexity and has one E1 voice port. There are also two E1 voice ports in HWIC slots. What is the maximum number of G.711 conferences that you can support with the extra DSPs?
Answer: The five PVDM-12s have three C549 DSPs each, which can terminate four medium-complexity calls requiring that 8 of the 15 DSPs are used for voice termination. You can configure the remaining seven DSPs to support one conference bridge each.
The two PVDM2-48s have a total of six C5510 DSPs. Four of these DSPs are required to terminate the 60 voice channels in the two E1s. The remaining two C5510 DSPs can each support eight G.711 conferences, for a total of 16.
Because using both Cisco IOS Media Resources (C549) and Enhanced IOS Media Resources (C5510) is not supported on a single router, you can register only the C5510s to CallManager. Therefore, the maximum number of G.711 conferences supported is 16.
Chapter 15
1 Why would you choose to implement an application in Tcl instead of VoiceXML?
Answer: A VoiceXML application requires an HTTP server, whereas a Tcl application does not require external servers.
2 Why would you choose to implement an application in VoiceXML instead of Tcl?
Answer: VoiceXML can interact with existing web applications requiring minimal development time to provide a voice interface to an existing web-based service. VoiceXML is also standards based, whereas Tcl IVR 2.0 uses Cisco proprietary APIs.
3 What is the precedence order for setting parameter values?
Answer: The precedence order is dial peer, service, and then package.
Answer: A package is a set of common functions that multiple applications use.
5 What is the purpose of a parameter namespace?
Answer: A parameter namespace allows multiple applications to use parameters that have the same name.
6 What application enhances the network management capabilities of Cisco routers?
Answer: Embedded Event Manager enhances the network management capabilities of Cisco routers.
7 What protocol can you use to increase the size and number of prompts available to an application?
Answer: You can use RTSP to increase the size and number of prompts available to an application.
8 What are the requirements for recording an application prompt?
Answer: You should record audio prompts in 8-bit, μ-law with 8-KHz encoding and save them in au format.
Chapter 16
1 List the two primary functions of a gatekeeper in an H.323 network.
Answer: Gatekeepers provide call routing with a centralized dial plan and CAC.
Answer: A gatekeeper zone is used to logically group H.323 devices that an individual gatekeeper controls.
3 What is a technology prefix?
Answer: A technology prefix is an optional H.323 feature that the gatekeeper uses to group gateways by type or class or to define a pool of gateways.
4 Where is bandwidth management initially done?
Answer: Bandwidth management is initially done when processing an ARQ. You can modify it later during the call with a BRQ.
5 What is another name for a directory gatekeeper?
Answer: Another name for a directory gatekeeper is a hierarchical gatekeeper.
6 What is GUP, and what is it used for?
Answer: GUP is the Gatekeeper Update Protocol. It uses RAS messages to provide updates to alternate gatekeepers when using gatekeeper clustering.
Chapter 17
1 What are two ways to verify that gateways have successfully registered with a gatekeeper?
Answer: Use the show gatekeeper endpoints command on the gatekeeper or the show gateway command on the gateway.
2 Why is it necessary to configure technology prefixes when implementing gatekeepers on a voice network?
Answer: Typically, a called number matches a zone prefix, not a registered E.164 address. In this case, if a technology prefix is not also matched, the call is rejected.
3 What do you need to do to the gatekeeper configuration to provide dynamic prefix registration support?
Answer: Add the rrq dynamic-prefixes-accept command to the gatekeeper configuration.
4 How much bandwidth does the gatekeeper count for each call using a G.729 codec? For a G.711 codec?
Answer: The gatekeeper counts each G.729 as 16 Kb and each G.711 call as 128 Kb.
5 How can you provide redundancy on a CallManager H.225 gatekeeper-controlled trunk?
Answer: You can provide redundancy on the trunk by assigning up to three subscriber servers to the CallManager redundancy group that is associated with the device pool assigned to the trunk.
6 How is state information shared between gatekeepers in a gatekeeper cluster?
Answer: State information is updated to all members of a gatekeeper cluster using GUP messages.
7 What three commands can provide detailed information about the status of a gatekeeper cluster?
Answer: The show gatekeeper status cluster, show gatekeeper cluster, and show gatekeeper zone cluster commands are available to verify the setup and status of the cluster.
8 Which undocumented trace command can you use to show the decision tree steps that the gatekeeper takes as it processes an ARQ message?
Answer: You can use the debug gatekeeper main 5 trace command to show the decision tree steps that the gatekeeper takes as it processes an ARQ message.
Chapter 18
1 Why would you use media flow-around?
Answer: You use media flow-around mode when you are not concerned with hiding your address. Because the IPIPGW is only involved in call setup and does not have to process media streams, more calls can be supported.
2 What is needed to implement an IPIPGW?
Answer: You need a supported router platform, an IPIPGW Cisco IOS image, and an integrated voice video feature license to implement an IPIPGW.
Answer: A via-zone is a local zone that has been dedicated to only service IPIPGWs. Although it is not necessary to have a dedicated zone for the IPIPGW, it is often much simpler to troubleshoot networks in which we have an dedicated IPIPGW zone.
4 Under what circumstances is a DSP needed in an IPIPGW call?
Answer: A DSP is needed if the two call legs do not use the same codec or packet size. If the call legs use identical parameters, a DSP is not required.
5 What platforms are supported for the Cisco Multiservice IPIPGW?
Answer: The supported platforms are the Cisco ISRs (Cisco 2800, Cisco 3800), Cisco 2600XM Series (Cisco 2691, 2650XM, 2651XM, 2620XM, 2621XM, 2610XM, and 2611XM), Cisco 3700 Series (Cisco 3725, 3745), Cisco 7200 VXR, Cisco 7301, Cisco AS5350XM, and Cisco AS5400XM.
6 Can the Cisco Multiservice IPIPGW register to more than one gatekeeper?
Answer: No, it can register with only one gatekeeper. However, more than one gateway can register to one gatekeeper.
7 Can you use the Cisco Multiservice IPIPGW without a gatekeeper?
Answer: Yes, you can. You can run the Cisco Multiservice without a gatekeeper by configuring direct dial peers on the gateway.
8 Are translation rules and digit manipulation supported on the Cisco Multiservice IPIPGW?
Answer: Yes, they have the same support for translation rules and digit manipulation on VoIP dial peers as other Cisco H.323 gateways.