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130 Cards in this Set
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Client-Server based Protocol implemented on Voice Gateways which development was greatly influenced by Cisco. The Gateway depends on a Call-Agent for intelligence of the Dial Plan.
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Media Gateway Control Protocol (MGCP)
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When a Gatekeeper needs to translate a telephone number to an IP Address of an endpoint located on another Gatekeeper, This RAS Message is sent.
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LocationRequest (LRQ)
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H.323 Gatekeepers route calls to endpoints located in different Local and Remote zones based on this.
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A "Zone Prefix"
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Eliminates the burden of a full-mesh topology between the Gatekeepers in an IP Network.
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Directory Gatekeeper
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Channel used for communicating with a Gatekeeper.
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Registration, Admission, Status (RAS)
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RAS Message sent when a Gateway wants to Register with the Gatekeeper but does not know its address.
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A GatewayReQuest (GRQ) Message is sent to the multicast address 224.0.1.41.
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Before a Gateway or Endpoint transmits H.225 Setup Messages to its destination, this message is sent to its Gatekeeper.
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AdmissionRequest (ARQ)
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These messages are only sent between Gatekeepers.
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LRQ, LRJ, LCF
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This RAS message is sent to the originating Gateway when the Gatekeeper is awaiting a response to an LRQ.
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A RequestInProgress (RIP) message
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This is the default mode for LRQ forwarding between redundant Gatekeepers.
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Sequential Forwarding. LRQs are sent to each GK in a prioritized list that match the Zone prefix. The process is terminated if an LCF is received by the original GK.
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Voice Gateways route calls to IP Addresses or Voice Ports based on configured...
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Dial Peers
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An Inbound Dial Peer configured to match destination-pattern 555.... will match an incoming call based on its DNIS or ANI?
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ANI. Inbound Dial Peers configured with destination-pattern match based on the Calling Number or ANI.
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Forwards LRQs to Gatekeepers matching the zone prefix in a single shot.
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Blast Forwarding. Opposite to Sequential Forwarding, which forwards LRQs to matching GKs one by one.
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This command is similar to "session target ipv4:xxx.xxx.xxx.xxx" for VoIP Dial Peers, but is used by gateways to talk to an H.323 Gatekeeper.
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"session target ras"
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Command set on gateway to advertise a hostname to its Gatekeeper upon registration.
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"h323-gateway voip h323-id GW1" will advertise GW1 as the gateways name to the Gatekeeper.
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This message is sent to a Gateway or endpoint when it requests for Admission from a gatekeeper it is not registered with.
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AdmissionRejected (ARJ)
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This command will set the maximum allowable bandwidth that can be used for calls in the NewYork zone and any other zone, to 2Mbits.
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bandwidth interzone zone NewYork 2000. from GK config.
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the bandwidth command for gatekeepers measures bandwidth capacity for zones in this unit.
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Kilobits Per Second (Kbps). Mbps specified with three consecutive 0s (i.e: 2Mbps = 2000)
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Feature of H.323 Version 2 where a TTL value is set during initial registration either in an RRQ or RCF message, to indicate how long to keep registration with the Gatekeeper active.
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H.323 Lightweight Registration.
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This happens when the TTL set during registration with a Gatekeeper is about to expire.
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The Keepalive value of the Registration Message is set to TRUE which refreshes the TTL.
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To keep registrations active, how often did Gateways re-register with their Gatekeeper prior to H.323 Version 2.
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Every 30-seconds gateways would register with their gatekeeper to maintain registration, this caused a lot of unnecessary overhead.
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For redundancy and call coverage multiple matching dial peers can be configured with this command to set priority in which Dial Peers are processed first. Also used to form Hunt Groups.
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preference [0,1,2,3,4,5,6,7]. the lower the preference the higher the priority of the dial peer.
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What is TEHO?
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Tail-End-Hop-Off: Toll bypassing technique which uses the WAN to reach a destination on the PSTN. A Call is routed to a site that is closest to the station on the PSTN.
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What are RAI RAS Messages
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RAI is short for Resource Availability Indicator.
With RAI, the Gateway alerts its Gatekeeper when DS0 or DSP channel resources become limited. |
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What is GKRCS?
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Gatekeeper-Routed Call Signaling. With GKRCS, Call Setup messages are proxied through the Gatekeeper rather than exchanged directly with another endpoint.
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What commands will create a translation profile and apply it to voice port?
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(config)# voice translation-rule <rule num>
(config)# voice translation-profile <name> (cfg-translation-profile)# translate <called/calling/redirect-called> <rule num> (config)# voice-port <port> (config-voiceport)# translation-profile <direction> <profile> |
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Cisco Unified Border Element (CUBE) can interoperate with what signaling protocols?
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SIP and H.323
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CUBE cannot inter operate with MGCP or SCCP
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True or False: Session Initiation Protocol or SIP was developed with a Client-Server Architecture.
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False. SIP is based on a Peer-to-Peer protocol architecture.
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SIP Clients (Endpoints) are known as this in SIP lingo
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User Agents Clients and User Agent Servers (UAC or UAS).
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How does one specify a telephone number instead of a USERID in a SIP Address?
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the user=phone parameter.
For example sip:5555555555@domain.com;user=phone |
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Mechanism used to prevent the loss of digits when voice packets are compressed.
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Dual-Tone Multifrequency Relay or DTMF Relay.
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Server which forwards messages to a UAS (SIP Server or other SIP UA) on behalf of a SIP UA.
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SIP Proxy Server.
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This server tells UAs the new IP Address to contact to reach a recently moved client.
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SIP Redirect Server
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Provides address resolution for SIP devices.
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SIP Location Server
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In this Media Mode, the IP Addresses and Port Numbers found in RTP headers of Voice Packets are replaced with that of the IPIPGW.
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Media Flow-Through Mode.
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In this Media Mode, IP Addresses and Port Numbers in Voice Packets are left untouched by the IPIPGW.
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Media Flow-Around Mode.
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True-or-False: Media Flow-Through is the default mode set on an IPIPGW.
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True. IPIPGWs default to Media Flow-Through for handling of RTP packets.
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True-or-False: Media Flow-Around provides privacy of endpoint IP addresses and can be used to hide the network topology.
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False: Media Flow-Around does not alter addresses found in the RTP header, thus endpoint addresses can be discovered. Media Flow-Through is the mode that provides privacy of addressing.
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What Cisco Component is used to Interconnect disparate networks that use different protocols for signaling Voice / Video?
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An IP-to-IP-Gateway or more specifically, a Cisco Unified Border Element
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You have configured the CUBE gateway to signal from an H.323 Network to an external network running SIP. After some investigation, you notice the SIP network is not able to communicate with the H.323 Network. How can you fix this?
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(config)# voice service voip
(config-voice-service)# allow-connections h323 to sip (config-voice-service)# allow-connections sip to h323. For a SIP and H.323 Network to communicate between each other you must enter the allow-connections command twice, indicating signaling is allowed from one connection to another and vice versa. |
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What commands enable VoIP protocol inter-networking on a Cisco Unified Border Element (IPIPGW).
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(config)# voice service voip
(config-voice-service)# allow-connections <proto1> to <proto2>. |
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Command to prevent sending both in-band and out-of-band DTMF digits between H.323 and SIP Dial Peers on an IPIPGW
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dtmf-relay rtp-nte digit-drop
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This protocol is responsible for re-timing and re-ordering voice packets so that they are played in the correct order to the listener.
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Real-Time Transport Protocol (RTP)
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What two key components found in RTP Headers contained in Voice Packets make it possible to properly re-order and playback a voice stream.
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Sequence Numbers and Timestamps. Sequence Numbers are used to only order packets, and are NOT used for retransmission purposes.
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True-or-False: If voice packets are lost, RTP re-transmits lost packets until an Acknowledgment is received.
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False: RTP packets are NEVER re-transmitted. Voice packets which are dropped are lost forever, unless using a sophisticated CODEC capable of reproducing the potential sample.
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Are RTP ports assigned with Static or Dynamic numbers?
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RTP does not use a pre-assigned port number(s), port assignment is done dynamically by devices using the protocol.
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The RTP Channels consist of one for replaying voice packets and one for reporting voice quality. Which one is assigned
Even port numbers and which uses Odd port numbers? |
RTP uses Even port numbers and is responsible for playing back packets. RTCP uses the next available odd port number from the one assigned for RTP and generates statistics for voice quality.
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What are two common Peer-to-Peer protocols used in Voice Over IP?
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Session Initiation Protocol (SIP) and H.323
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What are two common Client-Server protocols used in Voice Over IP?
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Media Gateway Control Protocol (MGCP) and Skinny Client Control Protocol (SCCP)
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Is MGCP (without PRI Backhauling) a TCP or UDP protocol? and what port is used?
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MGCP itself uses the User Datagram Protocol (UDP) port 2427 for exchanging messages.
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Protocol which transmits counters of packets sent/received, drops, errors, packet jitter and other statistics relating to overall voice quality.
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RTP Control Protocol. RTCP uses a separate channel from RTP that is responsible for exchanging statistics regarding the Quality of Service (QoS) of a call.
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How often are RTCP packets transmitted to an endpoint?
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RTCP packet transmissions default to Every 5 seconds
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What port range do Cisco devices use for their RTP (and RTCP) channels?
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RTP/RTCP ports on a Cisco devices range from 16384-32767
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Without using RTP Header Compression techniques (cRTP), what is the total packet size for the IP/UDP/RTP headers?
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IP, UDP, RTP headers combined together in a typical voice packet are 40-bytes in length.
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the headers for a VoIP packet combined together total to 40-bytes in length. Using cRTP to compress the headers, VoIP packets are reduced to how many bytes?
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packets compressed with cRTP are reduce VoIP packets down to 2 to 4 bytes in length
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True-or-False: cRTP should not be implemented on WAN links below T1 speeds.
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False: RTP header compression such as cRTP reduces bandwidth consumption of WAN links, and is ideal for slow speed WAN connections. T1 or higher speed links do not benefit from cRTP
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When a call is being set up between H.323 capable devices which protocol and port is used?
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Call Signaling is done via the H.225.0 protocol which uses TCP port 1720 as its defaults
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A gateway sends an ARQ message to its registered gatekeeper to place a call. what protocol and port will be used for this communication?
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UDP port 1719 is the default port used for unicast RAS communication.
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If a gateway sends a GRQ message to multicast address 224.0.1.41 to find a gatekeeper to register with on the network, what protocol and port is used?
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UDP port 1718 is the default port used for multicast RAS communication.
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GK1 has zones NYC, CHI and CAL configured. GK2 is configured with zones JPN and RUS. What type of Zones are JPN and RUS from GK1s perspective?
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JPN and RUS are Remote Zones to GK1
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When a Gatekeeper receives an AdmissionRequest (ARQ) from a gateway what does it attempt to match first?
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the Technology Prefix. The ARQ is first examined for a Technology Prefix contained in the dialed number. Then after the Zone Prefix is checked.
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a port identified by port 01/0:13 means what?
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the port is referencing a ds0-group with an ID of 13
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How many local zones should be configured on a Directory Gatekeeper?
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The DGK needs at least one Local Zone configured for other Gatekeepers to reference as a Remote Zone for forwarding LRQs to
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Wild-card used to forward all unknown zone prefixes to the Directory Gatekeeper
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Asterisk (*). GK1 can send an LRQ to the DGK zone by putting the command:
(config-gk)# zone remote DGK <domain> <ip/port optional> (config-gk)# zone prefix DGK * |
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What are the default values for cost and priority when deciding what gatekeeper to use when multiple LocationConfirmed (LCF) messages are received?
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Cost and Priority default to 50. When more than one Remote Gatekeeper responds to an LRQ, the Gatekeeper with selects the Remote Gatekeeper configured with either a Low Cost or High Priority when decided where to route the call
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zone remote <GK-NAME> <domain> <ip> cost <cost> priority <priority>
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what does the command gw-type-prefix 1#* default-technology do, and where is it configured?
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gw-type-prefix <prefix> default-technology is configured on the Gatekeeper and will route any unresolved telephone numbers to gateways registered with a 1#* technology prefix.
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what does the command gw-type-prefix 1#* gw ipaddr 192.168.1.1 do?
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Configured the Gatekeeper to recognize that RAS messages with a technology prefix of 1#* are to be routed to gateway 192.168.1.1
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What is a VIA-Zone?
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A VIA-Zone is a zone which contains a registered IPIPGW.
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True-or-False: CAMA trunks can only send outbound DNIS to a Public Safety Answering Point (PSAP)?
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False: CAMA trunks only support Outbound ANI (Calling Number) which the Public Safety Answering Point (PSAP) receives and processes
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What are two cards that support Centralized Automated Message Accounting (CAMA)?
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The VIC2-2FXO and VIC2-4FXO interface cards support CAMA via software configuration.
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What are some basic characteristics which make the Default Dial Peer (Dial Peer 0)?
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Dial Peer 0 consists of this configuration:
* Any Codec * IP Precedence 0 * VAD Enabled * No RSVP Support * fax-rate service POTS: * no ivr application |
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When a router matches an inbound call leg to an inbound dial peer, how are digits analyzed?
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the full string is matched (En Bloc matching) with Inbound Dial Peers
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When a router is matching a call leg to an outbound dial peer how are the digits analyzed?
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Routers match outbound call legs to outbound dial peers on a digit-by-digit basis, matching occurs as each digit is received by the router
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What are two popular DSP Chipsets found in modules for Cisco Routers?
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C549 and C5510 DSP chips
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True-or-False: High Complexity Codecs require less DSP resources
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False: High Complexity CODECS require more processing compared to Medium Complexity ones.
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What is the maximum number of voice calls and fax relay sessions that can be processed by a C549 DSP chip with Med and High Codec Complexity?
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C549 chips can process a maximum of 4 calls with Medium Complexity and 2 calls with High.
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What is the maximum number of voice calls and fax relay sessions that can be processed by a C5510 DSP chip with Med and High Codec Complexity?
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C5510 chip can process a maximum of 8 calls with Medium Complexity and 6 calls with High.
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What does the 'flex' complexity option do?
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Switch complexity modes to either Med or High based on the type of call that is made
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What is a disadvantage for using High Complexity codecs?
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Using one or more high compression Codec(s) require more DSP resources (processing) and results in fewer calls that can be processed by the DSP chips
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What command configures the codec complexity for DSPs?
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"codec complexity <complexity>" from voice-card configuration mode
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What are some Medium Complexity Codecs?
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G.711, G.726, G.729a, G.729ab and G.729a Annex B
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What are some High Complexity Codecs?
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G.728, G.723, G.729, G.729b, G.729 Annex B
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What is a PVDM2?
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The PVDM2 is short for Packet Voice Digital Signal Processor Module (PVDM). It is a SIMM Module which contains DSP chips for processing voice. PVDM2 modules come in different Module types with varying number of DSP chips.
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What are some common DSP Modules available and the number of DSPs found on them?
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PVDM2-8 = 1/2 DSPs
PVDM2-16 = 1 DSPs PVDM2-32 = 2 DSPs PVDM2-48 = 3 DSPs PVDM2-64 = 4 DSPs |
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True-or-False: the codec complexity flex command is available on DSP with the C549 chipset
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False: Flex complexity is only available for C5510 chipsets
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If DSPs are configured for FLEX complexity, what is the maximum call density?
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Sixteen voice calls per DSP
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What CODEC is available in both Medium and High Complexity modes?
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Fax Relay
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What Voice Codec is used for Fax Pass-Through communication?
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G.711 a-law and u-law
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Why are high compression codecs for Voice not suitable for Fax/Modem communication?
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High Compression standards such as G.729 are geared more towards encoding human speech patterns. If these codecs were used for Fax/Modem communication tones would become distorted and communication would fail.
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What is Codec Upspeed in Fax Pass-Through?
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Is a technique which can be employed so that when a gateway detects incoming tones used by a Fax, it immediately switches the voice codec to G.711, and disables Voice Activity Detection (VAD) and Echo Cancellation (EC) for the duration of the call until the fax is over.
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How much extra overhead is appended to Voice Packets when used with IPSec for calls over a VPN?
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An addition 50 - 57 bytes of overhead will be attached to every voice packet over IPSec
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This command will enter configuration of the D-Channel on a PRI interface
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(config)# interface serial 0/0:23
ISDN PRI connections have 23 Bearer (B) Channels and 1 Delta Channel for call signaling. The signaling channel is channel 23 for controllers on Cisco Routers because counting starts from 0 to 23 |
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What three elements are required for Fax Pass-Through to work?
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1) G.711, G.726/32 or a Clear-Channel Codec
2) VAD needs to be disabled 3) EC also needs to be disabled |
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What signaling protocols can be used with Fax Transmissions?
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H.323, SIP and MGCP
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What is Fax Pass-Through?
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Modulated Fax signals (Analog) are encoded with a PCM-based codec (i.e: G.711) and are transported end to end without any demodulation of the original signal
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Command to set the fax transmission speed of the gateway
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fax rate <speed>. Two standard speeds for fax transmission are 9600bps and 14.4Kbps
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Modem Relay uses this protocol for transporting Modem over the IP?
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SPRT. Modem Relay uses Simple Packet Relay Transport protocol (SPRT) which is a UDP protocol for delivery modem over IP
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Commands to manually enable Fax Pass-Through on a gateway
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(dial-peer-config)# fax protocol pass-through
(dial-peer-config)# no vad (voice-port)# no echo-cancel enable |
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What three methods are used to deliver Fax / Modem over IP?
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Pass-Through, Relay & Store-and-Forward
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True-or-False: T.38 Fax Relay is an industry standard protocol
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True: T.38 is an industry standard protocol developed by the developed by the International Telecommunications Union (ITU)
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This is the default protocol found on Cisco Devices for Fax Relay
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Cisco Fax Relay is set by default on Cisco Gateways and is Cisco Proprietary and not Industry Standard
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True-or-False: With Pass-Through employed, Gateways distinguish a Fax Call from a Voice Call
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False: Because Pass-Through does not alter the media and encodes Fax the same as Voice, Gateways can not tell the difference
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Fax Pass-Through requires how much bandwidth available?
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Because Fax Pass-Through uses a PCM codec like G.711 64Kbps is requried for transmission
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Command to set the protocol for Fax communication
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To set the protocol globally
voice service voip fax protocol <protocol> OR to set on an individual dial peer dial-peer voice <tag> voip fax protocol <protocol> |
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What two key components are required for CUBE to operate with RSVP-based Call Admission Control (CAC)?
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RSVP must be configured on two CUBE Gateways and configured with Media Flow-Through
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True-or-False: For RSVP to successfully operate with a CUBE, Media Flow-Around must be the configured flow type
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False: Media Flow-Around is not a supported flow type when dealing with RSVP-based CAC
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What is cRTP?
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cRTP is short for "compressed RTP" even though header 'suppression' takes place rather than compression. It is a feature which enables caching of redundant information sent during an RTP session
such as Source/Destinaion IP Address and Port Numbers. |
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What are some key features that make up Secure RTP (sRTP)
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* 128-bit Encryption of RTP Data
* Message Source Authentication * Message Integrity * Protection against Replay Attacks |
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What encryption standard is used by sRTP and what modes are supported?
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sRTP uses 128-bit AES Encryption and can operate in Segmented Integer Counter (SIC) or F8-Mode
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Which E&M feature group allows support for Outbound ANI?
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E&M FG-D EANA is the only feature group which supports Inbound and Outbound DNIS with Outbound ANI
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This command will tell the CUBE to let Codec Negotiation occur between the two endpoints and not to intervene.
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(dial-peer)# codec transparent
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Voice Interface common for Analog Switch and PBX connections
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Earth & Magneto (E&M)
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Which E&M Signaling Type is the most common used in North America?
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E&M Signaling Type I
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True-or-False: E&M Type IV is not supported on Cisco Gateways
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True. Instead E&M Type II can be used in place, as it operates the same as Type IV
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What two modes can E&M interfaces operate in?
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2 Wire or 4 Wire. Signaling can be used on one pair of wires (2-wire) or two pairs of wires (4-wire) depending on what the Telephony Equipment supports. The remaining wires are used to carry voice traffic.
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Which E&M Signaling Type is the most commonly used outside of North America?
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E&M Signaling Type V
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True-or-False: With E&M Wink-Start the originating side goes on-hook on its M-lead and waits for the Wink from the other end before sending address information
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False: The originating side goes off-hook on its E-lead and waits for the Wink to be received on its M-lead before sending address information
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What is the default sample size for packets of audio created by Cisco Devices, regardless of codec being used?
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20 milliseconds
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What command will display a list of configured Dial Peers as well as their operational status?
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# show dial-peer voice summary
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This command will display configuration pertaining to a specific Dial Peer
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# show dial-peer voice <tag>
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This command can be used to see what Dial Peer will be matched with the telephone number 5551212
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# show dialplan number 5551212
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What is the bandwidth required for calls using the G.729 codec?
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8000 Bits Per Second (bps) or 8Kbps
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What is the bandwidth required for calls using the G.723 codec?
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G.723 is available in either r53 (5.3Kbps) or r63 (6.3Kbps) bit-rate
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What will the command "zone prefix Zone_Y 555.... gw-priority 10 NewYork_GW" do if set on the Gatekeeper?
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When a call beginning with 555 reaches the Gatekeeper, it will first contact
NewYork_GW in Zone_Y before routing the call to any other Gateway in that zone. This is because NewYork_GW has been assigned the highest gateway priority (priority 10 is highest and 5 is the default). |
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What command is used to configure a gateway to register with a technology prefix?
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"h323-gateway voip tech-prefix <prefix>" set under the gateway interface
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As an alternative to configuring technology prefixes on the gateway, what command will tell the Gatekeeper the gateway to contact for calls destined for a given technology prefix?
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"gw-type-prefix <prefix> gw ipaddr <ip address of gateway>" from gatekeeper config
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This command will tell a gatekeeper to contact another gateway in the zone when resources become limited on a gateway
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"resource threshold high <value> low <value>" under gateway config
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What is the formula to calculate zone bandwidth for a gatekeeper?
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(Number of Calls) * (Codec Bandwidth) * 2 = Zone Bandwidth
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This command can be used to test translation rules and verify the result outputs as intended
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"test voice translation-rule <rule_num> <number>" can be used to simulate number translations before implementing into a production environment
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True-or-False: the Session Initiation Protocol (SIP) typically uses UDP ports
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False: SIP typically uses TCP or UDP ports 5060
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