voice

time it takes for speech to exit the speaker's mouth and reach the listener's ear.

length a signal must travel via light in fiber or electrical impulse in copper-based networks.

• is caused by devices that forward the frame through the network
• Includes queuing delay for congested networks.

amount of time it takes to actually place a bit or byte onto an interface

150 ms 500ms

• is a variation of packet interval time and only exist isn packet based networks
• the difference between when the packet is expected and when it is actually received is jitter

sends two 10-ms G.729 speech frames in every packet.

the 54-byte header, multiple voice samples can be packed into a single Ethernet frame to transmit

can increase the voice delay

Two basic variations of 64 Kbps PCM are commonly used: µ-law and a-law

encodes using 4-bit samples, giving a transmission rate of 32 Kbps

compression techniques that exploit redundant characteristics of the waveform itself

techniques that are grouped together

• 64kbps
• 40,32, 24 and 16
• 16 low delay
• 8 stream
• 5.3 and 6.3

free speech codec suitable for robust voice communication over IP. The codec is designed for narrow band speech

compare how well a particular codec works under varying circumstances, including differing background noise

developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.

developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.

loud and and long

packetloss, 20 ms of speach is average

5 percent of packet loss is average

hangover and is typically 200 ms.

beginning of a sentence is cut off or clipped or front end speach clipping

plague toll network

The network is designed to put all the dial-plan information in the central-site PBX.

simplifer the router configuration and use ios multimedia conference manager

RTP/UDP/IP

transmitting delay-sensitive traffic across packet-based networks

data part and a control part, the latter called RTP Control Protocol (RTCP).

• connectionless UDP protocol
• enables reliability without the need for a connection-based protocol such as TCP

example of joining disparate networks is when two companies merge.

provides a vehicle to identify

• sets the process of determining whether the client is
• allowed to perform or request certain tasks or operations

process of measuring resource consumption

• data-communications protocol designed to provide security management and statistics collection in
• remote computing environments

• •Transactions between the client
• and RADIUS server are authenticated through the use of a shared secret, which is never sent over
• the network.

• User passwords are sent encrypted between
• the client and RADIUS server

Each protocol has its own set of features and information fields

are the standards for every provider to offer billing-related information.

• include typical informational elements such as calling number, called number, connect time and
• date, call duration, and service characteristics.

contain data about which extension made or received calls to or from which number and for how long.

plain text log files

• access to user funds before the call is important so that you can see how much cash is available for
• the call to be made.

volume and mixed ussage and billing records

• •collects, correlates, and aggregates the
• accounting messages generated by the various VoIP-enabled network elements
• involved in a call.

• –It converts these into standard
• or proprietary CDR formats, such that one and only one CDR is generated for
• each call.

multiple technologies

• a third party should not be able to read the
• data that is intended for the recipient

• recipient should receive the packets that the originator sends without any change to their content. A
• third party should be unable to modify the packets in transit.

sender and recipient of VoIP signaling

protection from Denial-of-Service (DoS) attacks

Users share a single secret key

• Symmetrical. Same key used to
• encrypt and decrypt

Each user has a related public and private key

Asymmetrical. Different key used to decrypt than was used to encrypt.

hash is created using the original message

method to distribute public keys

issues a certificate validating the requestor’s identity and public key

–Evolved from SSL

• –Typically used to secure
• signaling

–Sits on top of TCP

The lower-level layer that provides connection security and is the workhorse.

• –It provides
• privacy and integrity.

• Transport mode—only the
• payload of an IP datagram is protected.
• Tunnel mode—, the entire IP packet is
• protected.

• provides integrity, authenticity, and privacy protection to the RTP traffic and to the control
• traffic for RTP,
• SRTP does not specify how the keys are exchanged between the sender and recipient

disable these unused services or ports for VoIP devices and IP infrastructure devices

secure critical voice devices such as call processing elements.

as a firewall between untrusted sources that send

All IP traffic on an untrusted port is blocked except for DHCP messages.

Can rate limit ARP requests to prevent flooding/DoS attacks.

• Configuring a maximum number of MAC addresses per port. If a particular port encounters this limit,
• the specified action is taken on that port.

Prevents malicious devices from sending STP BPDUs.

• monitor and analyze network traffic to detect intrusion
• Can be deployed on that VoIP side
• and on the data side of the IP network.

trust that is transmitted through another party.

h.323v5

• Also called endpoints, terminals provide
• point-to-point and multipoint conferencing for audio and, optionally, video and data

• interconnect to Public Switched Telephone Network(PSTN) or ISDN networks for H.323 endpoint
• interworking

provide admission control and address translation services for terminals or gateways.

• Devices that allow two or more
• terminals or gateways to conference with either audio and/or video sessions.

Uses the Gatekeeper Routed Call Signaling (GKRCS) model

• At the application layer
• Can manage QoS for a terminal that doesn’t support RSVP
• Can route H.323 traffic separate from data
• using application-specific routing (ASR)

most H.323 implementations today utilize TCP

Provides pre-call control in H.323 gatekeeper-based networks.

Used to connect, maintain, and disconnect calls between endpoints.

provides pre-call control

• multicast address is 224.0.1 .41
• Method for endpoints to determine which
• gatekeeper to register with

multicast message sent by an endpoint looking for the gatekeeper

The reply to an endpoint GRQ indicating the transport address of the gatekeeper’s RAS channel.

gatekeeper does not want to accept its registration.

• gatekeeper currently looks only at its static bandwidth table to determine whether to accept or reject the
• bandwidth request.

A reliable call control channel is created across an IP network on TCP port 1720

Call signaling messages are sent directly between two endpoints

Call signaling messages are routed through a gatekeeper

• controls the initiation, modification, and termination of interactive multimedia sessions.
• is a peer-to-peer protocol

• discover the location of the end user for the purpose of establishing a session or delivering a SIP
• request

enables the determination of the media capabilities

determination of the willingness of the end user to engagein communication.

parameters for the parties who are involved in the session.

the modification, transfer, and termination of an active session

logical function in the SIP network that initiates or responds to SIP transactions

A logical function that initiates SIP requests and accepts SIP responses

accepts SIP requests and sends back SIP responses

forwarding SIP requests to the target UAS or another proxy on behalf of the UAC.

that generates 300 class SIP responses to requests it receives

• accepts SIP REGISTER requests and updates the information from the request message into a location
• database.

entity that processes incoming SIP requests as a UAS.

• resolve host or domainnames into routable IP addresses. DNS can also be used to load-share across
• multiple servers in a cluster identified by a hostname.

describe the parameters of the multimedia session.

transports real-time data such as audio or video packets to the endpoints

reserve network resources such as bandwidth prior to establishment of the media session

provide privacy and integrity of SIP signaling information over the network

discover the presence any type of Network Address Translation (NAT) between them and the public Internet.

• identify a user or a resource within a network domain
• SIP URI sip:user@domain:portsip:user@host:port

globally routable and points to a domain whose location service can map the AOR to another SIP URI,

are sent from client to server to invoke a SIP operation

recipient user or service is invited to participate in a session

ACK request confirms that the UAC has received the final response to an INVITE request

capable of delivering a session to the user

the termination of a previously established session

enables UACs and network servers to cancel an in-progress request, such as INVITE

register its current location information corresponding to the AOR of the user with SIP servers.

Sent from server to client

The establishment of a session also results in a SIP signaling relationship between the peers

the establishment, modification, or termination of a media session.

• connection-oriented transport layer protocols such as TCP or Stream Control Transmission Protocol
• (SCTP)
• connectionless protocols such as UDP.

• providing centralized call control and feature management in SIP networks.
• initiate new SIP calls and modify and terminate existing calls.

UA needs the IP address of the registrar or proxy server to register and provide SIP service.

SIP endpoints register with a SIP registrar server.

are elements that route SIP requests to the UAS and SIP responses to the UAC.

that do not support the newer extensions, SIP defines the extension negotiation mechanism.

A SIP entity acts as a subscriber when it sends a SUBSCRIBE for a specific event type, such as message summary, to a SIP entity that the Request URI identifies.

The UAS that is processing the SUBSCRIBE request acts as the notifier

enables users to publish their availability status and display messages or icons as a form of self-expression