-
what is delay or latency
time it takes for speech to exit the speaker's mouth and reach the listener's ear.
-
what is propagation delay
length a signal must travel via light in fiber or electrical impulse in copper-based networks.
-
what is handling delay
- is caused by devices that forward the frame through the network
- Includes queuing delay for congested networks.
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what is serialization delay
amount of time it takes to actually place a bit or byte onto an interface
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What is one way delay for voip and total for satelite
150 ms 500ms
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what is jitter
- 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
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what is Packetizing Pulse Code Modulation
sends two 10-ms G.729 speech frames in every packet.
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what is Packetizing Pulse Code Modulation
the 54-byte header, multiple voice samples can be packed into a single Ethernet frame to transmit
can increase the voice delay
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what is voice compression
Two basic variations of 64 Kbps PCM are commonly used: µ-law and a-law
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adaptive differential pulse code modulation (ADPCM)
encodes using 4-bit samples, giving a transmission rate of 32 Kbps
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waveform codecs
compression techniques that exploit redundant characteristics of the waveform itself
-
source codecs
techniques that are grouped together
-
g.711 voice coding
.726
.728
.729
.723.1
- 64kbps
- 40,32, 24 and 16
- 16 low delay
- 8 stream
- 5.3 and 6.3
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iLBC (Internet Low Bitrate Codec)
free speech codec suitable for robust voice communication over IP. The codec is designed for narrow band speech
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Mean Opinion Score (MOS)
compare how well a particular codec works under varying circumstances, including differing background noise
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Perceptual Speech Quality Measurement
developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.
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Echo
developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.
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Echo has two drawbacks
loud and and long
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qos helps with what
packetloss, 20 ms of speach is average
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g.729 rule of thumb
5 percent of packet loss is average
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Voice Activity Detection (VAD)This fixed amount of time is what
hangover and is typically 200 ms.
-
detecting when speech begins
beginning of a sentence is cut off or clipped or front end speach clipping
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Digital-to-Analog Conversion
plague toll network
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Tandom Encoding
The network is designed to put all the dial-plan information in the central-site PBX.
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how to avoid Tandem Compression
simplifer the router configuration and use ios multimedia conference manager
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what are transport protocols
RTP/UDP/IP
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what kind of traffic does RTP send
transmitting delay-sensitive traffic across packet-based networks
-
RTP consists of
data part and a control part, the latter called RTP Control Protocol (RTCP).
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Reliable User Data Protocol (RUDP)
- connectionless UDP protocol
- enables reliability without the need for a connection-based protocol such as TCP
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Dial-Plan Design
example of joining disparate networks is when two companies merge.
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Authentication
provides a vehicle to identify
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Authorization
- sets the process of determining whether the client is
- allowed to perform or request certain tasks or operations
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Accounting
process of measuring resource consumption
-
Remote Authentication Dial-In User Service (RADIUS)
- 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
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Vendor-Specific Attributes (VSA)
Each protocol has its own set of features and information fields
-
call detail records (CDR)
are the standards for every provider to offer billing-related information.
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Automatic Messaging Accounting (AMA
- include typical informational elements such as calling number, called number, connect time and
- date, call duration, and service characteristics.
-
The VoIP network generates
CDRs that
contain data about which extension made or received calls to or from which number and for how long.
-
records stored in
plain text log files
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Prepaid Billing Applications
- 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.
-
2 voip challenges
volume and mixed ussage and billing records
-
Mediation Services
- •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.
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Security layering
multiple technologies
-
Confidentiality
- a third party should not be able to read the
- data that is intended for the recipient
-
Integrity
- 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.
-
Authenticity
sender and recipient of VoIP signaling
-
Availability
protection from Denial-of-Service (DoS) attacks
-
Shared Key
Users share a single secret key
- Symmetrical. Same key used to
- encrypt and decrypt
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Public-Key cryptography
Each user has a related public and private key
Asymmetrical. Different key used to decrypt than was used to encrypt.
-
Digital signature
hash is created using the original message
-
Certificates
method to distribute public keys
-
certificate authority (CA)
issues a certificate validating the requestor’s identity and public key
-
TLS
–Evolved from SSL
- –Typically used to secure
- signaling
–Sits on top of TCP
-
TLS record protocol
The lower-level layer that provides connection security and is the workhorse.
- –It provides
- privacy and integrity.
-
IPSec 2 different modes
- Transport mode—only the
- payload of an IP datagram is protected.
- Tunnel mode—, the entire IP packet is
- protected.
-
SRTP
- 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
-
Disabling Unused Ports/Services
disable these unused services or ports for VoIP devices and IP infrastructure devices
-
HIPS
secure critical voice devices such as call processing elements.
-
DHCP Snooping
as a firewall between untrusted sources that send
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IP Source Guard
All IP traffic on an untrusted port is blocked except for DHCP messages.
-
Dynamic ARP Inspection (DAI)
Can rate limit ARP requests to prevent flooding/DoS attacks.
-
CAM Overflow and Port Security
- Configuring a maximum number of MAC addresses per port. If a particular port encounters this limit,
- the specified action is taken on that port.
-
BPDU Guard and Root Guard
Prevents malicious devices from sending STP BPDUs.
-
NIPS
- monitor and analyze network traffic to detect intrusion
- Can be deployed on that VoIP side
- and on the data side of the IP network.
-
Transitive trust
trust that is transmitted through another party.
-
latest version of h.323
h.323v5
-
Terminals
- Also called endpoints, terminals provide
- point-to-point and multipoint conferencing for audio and, optionally, video and data
-
Gateways
- interconnect to Public Switched Telephone Network(PSTN) or ISDN networks for H.323 endpoint
- interworking
-
Gatekeepers
provide admission control and address translation services for terminals or gateways.
-
Multipoint control units (MCU)
- Devices that allow two or more
- terminals or gateways to conference with either audio and/or video sessions.
-
Call Control Signaling
Uses the Gatekeeper Routed Call Signaling (GKRCS) model
-
Proxy Server
- 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)
-
H.323 Protocols
most H.323 implementations today utilize TCP
-
Registration, Admissions and Status (RAS) Signaling
Provides pre-call control in H.323 gatekeeper-based networks.
-
Call control signaling
Used to connect, maintain, and disconnect calls between endpoints.
-
RAS Signaling
provides pre-call control
-
Gatekeeper Discover
- multicast address is 224.0.1 .41
- Method for endpoints to determine which
- gatekeeper to register with
-
Gatekeeper Request (GRQ)
multicast message sent by an endpoint looking for the gatekeeper
-
Gatekeeper Confirm (GCF)
The reply to an endpoint GRQ indicating the transport address of the gatekeeper’s RAS channel.
-
Gatekeeper Reject (GRJ)
gatekeeper does not want to accept its registration.
-
Bandwidth Control
- gatekeeper currently looks only at its static bandwidth table to determine whether to accept or reject the
- bandwidth request.
-
Call Control Signaling
A reliable call control channel is created across an IP network on TCP port 1720
-
Direct Endpoint Call Signaling
Call signaling messages are sent directly between two endpoints
-
GKRCS
Call signaling messages are routed through a gatekeeper
-
Session Initiation Protocol (SIP)
- controls the initiation, modification, and termination of interactive multimedia sessions.
- is a peer-to-peer protocol
-
User Location
- discover the location of the end user for the purpose of establishing a session or delivering a SIP
- request
-
User Capabilities
enables the determination of the media capabilities
-
User Availability
determination of the willingness of the end user to engagein communication.
-
Session Setup
parameters for the parties who are involved in the session.
-
Session Handling
the modification, transfer, and termination of an active session
-
User Agent (UA)
logical function in the SIP network that initiates or responds to SIP transactions
-
User Agent Client (UAC)
A logical function that initiates SIP requests and accepts SIP responses
-
User Agent Server (UAS)
accepts SIP requests and sends back SIP responses
-
Proxy
forwarding SIP requests to the target UAS or another proxy on behalf of the UAC.
-
Redirect Server
that generates 300 class SIP responses to requests it receives
-
Registrar
- accepts SIP REGISTER requests and updates the information from the request message into a location
- database.
-
Back-to-back user agent (B2BUA)
entity that processes incoming SIP requests as a UAS.
-
DNS
- 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.
-
Session Description Protocol (SDP
describe the parameters of the multimedia session.
-
Realtime Transport Protocol (RTP)
transports real-time data such as audio or video packets to the endpoints
-
RSVP
reserve network resources such as bandwidth prior to establishment of the media session
-
TLS
provide privacy and integrity of SIP signaling information over the network
-
STUN
discover the presence any type of Network Address Translation (NAT) between them and the public Internet.
-
SIP Addresses
- identify a user or a resource within a network domain
- SIP URI sip:user@domain:portsip:user@host:port
-
Address-of-Record (AOR).
globally routable and points to a domain whose location service can map the AOR to another SIP URI,
-
SIP Request Messages
are sent from client to server to invoke a SIP operation
-
INVITE
recipient user or service is invited to participate in a session
-
ACK
ACK request confirms that the UAC has received the final response to an INVITE request
-
OPTIONS
capable of delivering a session to the user
-
BYE
the termination of a previously established session
-
CANCEL
enables UACs and network servers to cancel an in-progress request, such as INVITE
-
REGISTER
register its current location information corresponding to the AOR of the user with SIP servers.
-
SIP Response Messages
Sent from server to client
-
Dialogs
The establishment of a session also results in a SIP signaling relationship between the peers
-
transaction
the establishment, modification, or termination of a media session.
-
Signaling SIP transactions
- connection-oriented transport layer protocols such as TCP or Stream Control Transmission Protocol
- (SCTP)
- connectionless protocols such as UDP.
-
B2BUA (Back-to-back user
agent)
- providing centralized call control and feature management in SIP networks.
- initiate new SIP calls and modify and terminate existing calls.
-
User Agent Discovering SIP Servers in a Network
UA needs the IP address of the registrar or proxy server to register and provide SIP service.
-
SIP Registration Process
SIP endpoints register with a SIP registrar server.
-
SIP Proxies
are elements that route SIP requests to the UAS and SIP responses to the UAC.
-
SIP Extensions
that do not support the newer extensions, SIP defines the extension negotiation mechanism.
-
SUBSCRIBE
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.
-
NOTIFY
The UAS that is processing the SUBSCRIBE request acts as the notifier
-
Presence
enables users to publish their availability status and display messages or icons as a form of self-expression
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