Cvoice

642-436 CVOICE 6.0 Exam Topics (Blueprint)

Exam Description
The 642-436 Cisco Voice over IP (CVOICE) is the exam associated with the Cisco Certified Voice Professional CCVP certification. This exam tests a candidate's knowledge of the foundational elements of VOIP calls, and the description of dial plans, and the implementation of gateways, gatekeepers and IP-IP gateways. Candidates can prepare for this exam by taking the CVOICE Cisco Voice over IP course.


Exam Topics
The following topics are general guidelines for the content likely to be included on the Remote Access exam. However, other related topics may also appear on any specific delivery of the exam. In order to better reflect the contents of the exam and for clarity purposes, the guidelines below may change at any time without notice.


Describe the components of a gateway
Describe the function of gateways

Describe DSP functionality

Describe the different types of voice ports and their usage

Describe dial peer types

Describe codecs and codec complexity


Describe a dial plan
Describe a numbering plan

Describe digit manipulation

Describe path selection

Describe calling privileges

Describe call coverage


Describe the basic operation and components involved in a VoIP call
Describe VoIP call flow

Describe RTP, RTCP, cRTP, and sRTP

Describe H.323
Describe MGCP

Describe SCCP

Describe SIP

Identify the appropriate gateway signaling protocol for a given situation

Describe voice quality considerations

Choose the appropriate codec for a given situation


Implement a gateway
Describe the gateway call routing process

Configure analog voice ports

Configure digital voice ports

Describe considerations for PBX integration

Configure dial-peers

Configure hunt groups and trunk groups

Configure digit manipulation

Configure calling privileges

Verify dial-plan implementation

Implement fax and modem support on a gateway

Configure a gateway to provide DTMF support


Describe the function and interoperation of gatekeepers within an IP Communications network
Describe the function and types of gatekeepers

Describe the interoperation of devices with a gatekeeper

Describe gatekeeper signaling

Describe Dynamic Zone Prefix Registration with a gatekeeper

Describe gatekeeper redundancy


Implement a gatekeeper
Configure devices to register with a gatekeeper

Configure gatekeeper to provide dial-plan resolution

Configure gatekeeper to provide call admission control

Verify gatekeeper operation


Implement an IP-to-IP gateway
Describe the IP-to-IP gateway features and functionality

Configure gatekeeper to support an IP-to-IP gateway

Configure IP-to-IP gateway to provide address hiding

Configure IP-to-IP gateway to provide protocol and media interworking

Configure IP-to-IP gateway to provide call admission control

Verify IP-to-IP gateway implementations

Describe H.323

H.323 is a open standard protocol suite that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
H.323 is designed to use IP at Network layer and has its own protocols defined for upper layers.

When we pickup a phone and setup a call H.225.0 is used and call is connected. It is responsible for ringing phone at other side.

Once phone is picked up H.245 is used to negotiate the features, like codec to be used for this calls. If codec negotiation fails both side will receive busy signal.

Once codec and other features have been negotiated optionally H.225 RAS (Registration, Admission and Status) is used to send call status to a gatekeeper.
Gatekeeper is used to keep track of all live calls in network. we will discuss this further in Gatekeeper section.

Next optional component is RTCP (Real Time Control Protocol) which s used to gather statistics on quality aspects of the media distribution during a session and transmit this data to the session media source and other session participants.

H.323 has protocols for video and data transfer as shown in diagram above. Example for use of these protocols is Netmeeting.


H.323 Components

H.323 Terminal
A H.323 terminal is an endpoint in the LAN that participates in real-time, two-way communications with another H.323 terminal, gateway, or multipoint control unit (MCU). A terminal must support audio communication and can also support audio with video, audio with data, or a combination of all three.
Cisco IP phone is not a H.323 terminal.

H.323 GateWay
H.323 gateway is a device required to make calls to PSTN, IP to IP and other H.323 network.
IP to IP gateway is used to make calls between network belong to different administrator groups.

H.323 Gatekeeper
H.323 Gateway manage availability of bandwidth in network to make calls. They are not required for small network. We create Zone of control to be managed by a gatekeeper. Gateway approach Gatekeeper and they redirect them to correct gateway and reserve bandwidth for the call duration. More on it in Gatekeeper section.

H.323 MCU (Multipoint Control Unit)
H.323 MCU primarily handles conference call, they collect media stream from all devices and send it back to all of them as a single stream. It could be a router of DSP resources on router.

H.323 call setup
Cisco router use H.323 by default if we have to use MGCP/SIP we have to configure them.
When we pickup phone, we hear dial tone dial the number.
H.225 is used to send signal to router and further from router to another phone. Second phone after getting sin gal from router start ringing.
Once phone is picked up H.245 is used to negotiate features, if features (codec) successfully are negotiated call is setup and you can talk if not both phones hear busy tone.
RTP is used to transfer audio stream between the phones.
With RTP, after every small interval RTCP transfer statistical data between phones.


If there is a gatekeeper in network all gateway will register with gatekeeper by RRQ (Registration Request), Gatekeeper will reply with RCF (Registration confirmation) msg or RRJ ( Registration Reject) msg.

When a phone dial number, gateway matches dial peer, which may redirect to check with Gatekeeper. Gateway sends ARQ (Admission Request) msg Gatekeeper looks into database and availability of bandwidth between gateways and replies with ACF/ARJ msg.
RRQ,RRJ,RCF,ARQ,ARJ,ACF all are part of H.225 RAS protocol.
Gateway send H.225 signals to gateway suggested by Gatekeeper, receiving gateway approach Gatekeeper before processing call to authorize call.
Once authorised by Gatekeeper, H.245 signal is used to negotiate codec/features and information is passed to Gatekeeper.
Once call is ended DRQ (Disengage Request) msg is send to Gatekeeper to update its database. Gatekeeper replies with DCF msg to both gateways and updates it database.