Audio Codecs Video Codecs

G.711 G.728

G.722 G.729

G.723.1 G.726

H.261

H.263

H.264 MPEG-4

MGCPMedia GatewayControl Protocol

SGCPSimple GatewayControl Protocol

H.248MEGACO

Media GatewayControl

GCP SIP

SDPSession

DescriptionProtocol

SAPSession

AnnouncementProtocol

H.323

SIP SessionInitiationProtocol

H.225.0RAS

RegistrationAdmission

Status

H.245Call Control

H.225.0Q.931

Call SetupRTSP

Real TimeStreamingProtocol

RTCPReal Time

ControlProtocol

NCSNetwork-BasedCall Signaling

Other

Cisco SCCPSkinny Client

ControlProtoc ol

TCP/UDPUDP/RUDP

Media Control SignalingApplications

SkypeSignaling

TCP/UDP

SCTPStream ControlTransmission

Protocol

cRTPCompressed Real Time Protocol

RTPReal TimeTransportProtocol

H.235Security

IPDCIP Device

Control

AAA

IPDRInternet Protocol

Detail Record

OSPOpen Settlement

Protocol

Real-timeConferencing

T.120

FoIPFacsimile over IP

T.38

H.450Supplement

Services(e.g.,Call Waiting)

MoIPModem over IP

V.150

Services

RADIUSRemote Authenticaion Dial-In User Service

IP

VOIP Quick GuideVOIP Quick Guide

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VOIP Architecture

VOIP Protocols

Authentication Authorization Accounting

PSTN

Internet

LANLAN

Gateway

WAN

MediaGateway

SoftSwitch

H.323Gatekeeper

Gateway

Gateway

Router

Router

H.323Gatekeeper

H.324Multimediaover POTS

H.320Multimediaover ISDN

GCPPhone

GCPTerminal

GCPPhone

SIPTerminal

SIPTerminal

SIPPhone

Router

H.323Phone

H.323Terminal

H.323Phone

H.323Terminal

SIPServer

Gateway

Router

Class 5Switch

SS7 Signaling

GCP Signaling

SIP Signaling

H.323 Signaling

RTP Traffic

PSTN Traffic

ISDNPhone

POTSPhone

LANLAN

MSUMediaGateway

SignalingGateway

SoftSwitch

Router

Router

H.323 Network Architecture

Gateway Control Protocol (GCP)Network Architecture

Session Initiation Protocol (SIP)Network Architecture

SIPServer

Router

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VOIP Technology Comparison

Standards body

Architecture

Call control

Endpoints

Signaling transport

Multimedia

Media transport

DTMF-relay transport

Fax-relay transport

Supplemental services

H.323

ITU-T

Distributed

Gatekeeper

Gateway, terminal

TCP/UDP

Yes

RTP-Real Time Transport Protocol

RTP

T.38

By endpoints or call control

SIP

IETF

Distributed, Peer-to-Peer

Proxy/Redirect Server

User agent

TCP/UDP

Yes

RTP-Real Time Transport Protocol

RTP

T.38

By endpoints or call control

MGCP/H.248/Megaco

MGCP/Megaco by IETF; H.248 by ITU-T

Centralized

Call agent/Media Control Gateway / Softswitch

Media Gateway, dump terminal

MGCP - UDP H.248/Megaco-TCP/UDP

Yes

RUDP - Reliable User Datagram Protocol; RTP - Real Time Transport Protocol

RTP

T.38

By call agent

H.323 - A Distributed VOIP NetworkH.323 Network Elements

Terminals: a network endpoint which may provide audio, data and video, communicat ions with another H.323 terminal. Gateways: a network funct ion that provides access to terminals on a circuit switched network (such as the PSTN) or another H.323 network. Gatekeepers: a network funct ion that provides address translat ion, access control, bandwidth management, and other management operat ions.Multipoint Control Units: a network funct ion that allows three or more terminals to part icipate in a mult ipoint conference.

H.323 gatewayH.323 gatkeeper

H.225 TCP connect ion

SETUP

CONNECT (H.245 address)

H.245 TCP connect ion

Capabilit ies exchange

RTCP address

CONNECT (H.245 address)

RTCP addresses

RTCP&RTP addresses

RTP stream

RTP stream

H.323 gateway IP phone

H.245signaling

H.225signaling

Example H.323 Call Flow ITU-T H.323 Standards

Standard#

H.323

H.225.0

H.235

H.239

H.245

H.246

H.350

H.360

H.450

H.460.x

H.501

H.510

Description

For secur ity in H.323 network.

H.323/PSTN Interworking.

Supplements in H.323.

Mobility for H.323 multimedia systems.

An umbrella recommendation of ITU-T thatdefines the protocols to provide audio-visual communication sessions on any packet network.

For call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats.

For dual stream use in videoconferencing.

A control protocol for multimedia communication.

Directory Services Architecture for Multimedia Conferencing.

An architecture for end-to-end QoS control and signaling.

For supplementary services such as call waiting, call forwarding, etc.

Protocol for mobility management and intra/inter-domain communication in multimedia systems.

SIP: Session Initiation Protocol A Peer-to-Peer VOIP Network

SIP Network Elements

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User Agents: A software program installed in a user’s terminal or an IP phone to initiate and terminate phone calls, plus data and video communications. There are two logic parts in the user agents: User Agent Server (UAS) and User Agent Client (UAC). UAC sends requests and receives responses. UAS receives requests and sends responses. Proxy Server: Performs routing of a session invitations according to invitee's profile. There are two basic types of SIP proxy servers--stateless and stateful. Stateless servers are simple message forwarders. Stateful proxies, upon reception of a request, create a state and keep the state until the transaction finishes.Redirect Server: Receives a request and sends back a reply containing a list of the current location of a particular user, by looking up the intended recipient of the request in the location database created by a registrar.Registrar Server: A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain in handles.

Example SIP Call Flow IETF SIP Standards

RFC#

2974

2976

3262

3263

3265

3311

3313

3327

3329

3420

3428

3486

4028

4168

4412

4566

3261

Description

Session Announcement Protocol (SAP)

The SIP INFO Method

SIP: Session Initiation Protocol (updated by RFC 3853, RFC 4320)

Reliability of Provisional Responses in SIP

SIP: Locating SIP Servers

SIP-Specific Event Notification

SIP UPDATE Method

Private SIP Extensions for Media Authorization

SIP Extension for Registering Non-Adjacent Contacts

Security Mechanism Agreement for SIP Sessions

Internet Media Type message/sipfrag

SIP Extension for Instant Messaging

Compressing SIP

Session Timers in SIP

SCTP as a Transport for SIP

Communications Resource Priority for SIP

SDP: Session Description Protocol

SIP Phone A

RTP/RTCP stream

SIP PROXY SIP Phone B

SIP/SDP INVITESIP/SDP INVITEStatus:100 Trying

Status:183 Session Progress

Status:183 Session Progress

Status:200 OKStatus:200 OK

SIP ACKSIP ACK

SIP:BYESIP:BYE

Status:200 OKStatus:200 OK

GCPs: Gateway Control Protocols A Centralized VOIP Network

Media Gateway Control Protocol (MGCP) and H.248/MEGACO Network ElementsMedia Gateway Controller(MGC): also known as Call Agent or Softswitch, it controls a number of dumb terminals and Media Gateways. The MGC receives signaling information from MG and can instruct it to alert the called party, to send and receive voice data etc.Media Gateway(MG): acts as a translation unit between disparate telecommunications networks such as PSTN, IP Networks, Mobile access networks or PBX.Signaling Gateway (SG): A component responsible for translating signaling messages between IP network and PSTN.Endpoints: Provide audio, data and video communications with another GCP terminal or a PSTN phone via gateway.

1

2

3

4

5

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Example GCP Call Flow MGCP Documents

CallAgent

MGCPGat eway A

Endpo in t A

MGCPGat eway B

Endpo in t B

RQNTRQNT

RQNT Response

NTFY f rom A

CRCX

CRCX Response

MDCX

MDCX Response

CRCX

CRCX ResponseRinging

Answer&

RQNT Response

RTP

RTP

RTCP

NTFY f rom A

DLCX

DLCX Response DLCX Response

DLCX

VoI P

OnHook

Off Hook& Dia ll ing

Med ia Gat eway Cont ro l Pro t oco l (MGCP) Version 1 . 0

Basic MGCP Packages

MGCP Ret urn Code Usage

MGCP CAS Packages

MGCP Business Phone Packages

MGCP Red irect and Rese t Package

MGCP Lockst ep St a t e Report ing Mechan ism

Med ia Gat eway Cont ro l Pro t oco l Arch it ect u re and Requ irement s

RFC 3435

RFC 3660

RFC 3661

RFC 3064

RFC 3149

RFC 3991

RFC 3992

RFC 2805

Standard# Descri pti on

H.248/Megaco Standards Main difference between Megaco/MGCP

H.248/Megaco version 1

H.248/Megaco Version 2

H.248/Megaco Version 3

H.248.1v1

H.248.1v2

H.248.1v3

RFC 3525

RFC 3054

Gateway Control Protocol Version 1

Megaco IP Phone Media Gateway Applicat ion Prof ile

Standard ITU-T File IETF File Description Megaco/H.248

A call is represented by teminat ions within a call context

Call types include any combinat ion of mult imedia and conferencing

Syntax is text binary

Transport layer is TCP or UDP

Defined by the IETF and ITU

A call is represented by endpoints within connect ions

Call types include point-to-point and mult ipoint

Syntax is text

Transport layer is UDP

Defined by Cisco and circulated in IETF

MGCP

VOIP Media Transport Protocols

Real-Time Transport Protocol (RTP)

Real Time Control Protocol (RTCP)

Real Time Streaming Protocol (RTSP)

Reliable User Datagram Protocol (RUDP)

Secure Real-t ime Transport Protocol (SRTP)

Stream Control Transmission Protocol (SCTP)

ZRTP

For delivering audio and video over the Internet.

Provides out-of-band control information for an RTP f low.

For use in streaming media systems which allows a client to remotely control a streaming media server and allowing t ime-based access to f iles on a server.

Used as the transport protocol for MGCP based network.

Provides encrypt ion, message authent icat ion and integrity, and replay protect ion to the RTP data in both unicast and mult icast applicat ions.

Provides transport services to ensure reliable, in-sequence transport of messages with congest ion control.

An extension to RTP which describes a method of Diff ie-Hellman key agreement for SRTP.

RFC 3550

RFC 3550

RFC 2326

RFC 1151

RFC 3711

RFC 2960

Draft

Functions Standard#Protocol Name

Class-Based Weigh t ed Fa ir Queu ing (CB-WFQ)

Cust om Queu ing (CQ)

Fa ir Queu ing (FQ)

Prio r it y Queu ing(PQ)

Prio r it y Queu ing - Class-Based Weigh t ed Fa ir Queu ing

Weigh t ed Fa ir Queu ing (WFQ)

Weigh t ed Random Early Drop / Det ect (WRED)

Type o f Service (ToS)

Diff Serv

I n t Serv

Po licy-based Rout ing

Resource Reserva t ion Pro t oco l (RSVP)

Commit t ed Access Rat e (CAR)

Generic Tra ff ic Shap ing(GTS)

Mult i-Class Mu lt i l ink Po in t - t o -Po in t Pro t oco l (MCML PPP)

Frame Re lay Forum 12 (FRF. 12)

Maximum Transmission Un it (MTU)

Compressed Rea l Time Transport Pro t oco l (cRTP)

Random early de t ect ion o r Random early d iscard (RED)

CategoryQueuing

Packet Classification

Traffic shaping and policing

Fragmentation

Other

Technology

WMV Windows Media Video

Also known as H. 264 o r AVC, i t is used f o r in t e rne t , b roadcast , and on st o rage med ia .

H. 261

H. 263

H. 264

MPEG-4 Part 2

MPEG-4 Part 10

DivX

X264

I TU-T version o f MPEG-4 Part 10

MPEG

MPEG

Based on MPEG-4 Part 2

Microso f t

Based on H. 264 ; GPL

Used p r imarily in o lder video con f e rencing and video t e lephony p roduct s.

Used p r imarily f o r videocon f e rencing , video t e lephony, and in t e rne t video .

Also known as MPEG-4 Part 10 , o r AVC (f o r Advanced Video Cod ing).

Used f o r in t e rne t , b roadcast , and on st o rage med ia .

Used f o r in t e rne t , b roadcast , and on st o rage med ia .

I t can do anyt h ing f rom low reso lu t ion video f o r d ia l up in t e rne t users t o HDTV.

A GPL-l icensed imp lement a t ion o f H. 264 encod ing st andard .

I TU-T

I TU-T

,

5 - , 4 - , 3 - and 2

G. 711

G. 721

G. 722

G. 722 . 1

G. 722 . 2

G. 723

G. 723 . 1

G. 726G. 727G. 728G. 729

Speex

iLBC (Internet Low Bitrate Vocoder)

L16

U-law (US, Japan) and A-law (Europe) compand ing .

Rep laced by G. 726 .

Subband-codec t ha t d ivides 16 kHz band in t o t wo subbands, each codedusing ADPCM.

Superceded by G. 726 ; Th is is a comple t e ly d iff e ren t codec t han G. 723 . 1

Part o f H. 324 video con f e rencing .

Rep laces G. 721 and G. 723 . Re la t ed t o G. 726 .

VOI P App lica t ions.

VOI P

VOI P

AMR-WB is st andard ized f o r usage in ne t works such as UMTS.

I TU-T

I TU-T

I TU-T

I TU-T

I TU-T

I TU-T

I TU-T

I TU-TI TU-TI TU-T

Freeware

I ETF RFC 3951Freeware

I ETF RFC 3551

PCM

ADPCM

ADPCM

Transf o rm-based

AMR-WB

DPCM

MPC-MLQ or ACELP

ADPCMADPCMLDCELPCS-ACELP

CELP

LPC

LPC

Uncompressed audio data samples

64

32

64

24/ 326.60, 8.85, 12.65, 14.25,15.85, 18.25, 19.85, 23.05 and 23.85

24 / 40

5 . 6 / 6 . 3

16 / 24 / 32 / 40

168

8, 16 , 32

8 , 16

8 , 16

128

8

8

16

16

16

8

8

888

2.15-24.6 (NB)

4-44.2 (WB)

13 . 3

15 . 2

Variab le

0 . 125

Sampling

Sampling

20

Sampling

Sampling

30

0. 125Sampling0 . 62510

30 ( NB ) 34 ( WB )30

20

Sampling

0 . 75

40

30

30

1

0. 62515

30 34–

30

20

4. 1

3 . 7 -3 . 9

3 . 9

3 . 63 . 9

4 . 0

4 . 0

Codec Type

Standard by

Modulation method

Bit rate (kb/s)

Sampling rate (kHz)

Frame size (ms)

Compression delay

Mean Opinion Score (MOS) Notes

Audio CODECs

CODEC Name ApplicationsStandard

by

Video CODECsVOIP CODECsQoS Technologies

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VOIP GlossaryACELP--Algebraic Code Excited Linear PredictionADPCM--Adaptive Differential Pulse Code Modulation AMR-WB--Adaptive Multi Rate – WideBand ATA--Analog Telephone Adaptor connects the conventional telephone to the Internet, converts the analog voice signals into IP packets, delivers dial tone and manages the call setup.Broadband--High speed Internet connection, such as cable TV, DSL or dedicated telecom lines(T1/E1).C7--Common Channel Signaling 7.Cable modem--A device used to connect a computer to the high speed coaxial cable run by cable TV companies to provide access to the Internet.CALEA--Communications Assistance for Law Enforcement Act.Call agent--The intelligent and controlling entity in an MCGP based IP telephony network.Call flow--The setup and tear down process and steps for a call to start till finish.CB-WFQ--Class-Based Weighted Fair Queuing.CDR--Call Detail Record.CID--Caller Identification (ID).Circuit switched network--The traditional telephone network used for making phone calls since 1878.Codec--Compressor-Decompressor or enCOder/DECoder process.Committed Access Rate (CAR)--A QoS feature.Compression--The squeezing of data in a format that takes less space to store or less bandwidth to transmit.Compression delay--The delay caused by the compression of data.Congestion--The situation in which the traffic present on the network exceeds available network bandwidth/capacity.CoS--Class of Service.CPE--Customer Premises Equipment.cRTP--Compressed Real Time Transport Protocol.CS-ACLEP--Conjugate-Structure Algebraic-Code-Excited Linear-Prediction. Custom Queuing-- A queuing method that allows a customer to reserve a percentage of bandwidth for specified protocols. Data compression-- The process to compress large data files into small files so that they use less bandwidth during transmission and less disk space when stored.Decompression--Process by which the full data content of a compressed file is restored.DiffServ-- An architecture for implementing scalable service differentiation in the Internet for QoS.DivX-- A video codec used for internet, broadcast, and on storage media.DPCM--Differential Pulse Code Modulation. DSL modem-- A device used to connect computers to the DSL line provided by a DSL operator to gain access to the Internet.DTMF--Dual-Tone Multi Frequency.Dynamic Jitter Buffer-- Collects voice packets, stores them, and shifts them to the voice processor in evenly spaced intervals to reduce any distortion in the sound.E&M (Ear and Mouth)--A type of supervisory line signaling. E911--Enhanced 911; used for providing emergency service on cellular and Internet voice calls.Emergency 911 calls--An emergency telephone number that handles all calls related to police, fire or medical emergencies in North America.Fair Queuing--A scheduling scheme to allow several data flows to fairly share the link capacity.FoIP--Fax over Internet Protocol.Frame Relay Forum 12 (FRF.12)--A Frame Relay specification of fragmenting Frame Relay frames into smaller frames.G.711--ITU-T specification of audio CODEC. G.721--ITU-T specification of audio CODEC.G.722--ITU-T specification of audio CODEC. G.722.1--ITU-T specification of audio CODEC.G.722.2--ITU-T specification of audio CODEC. G.723--ITU-T specification of audio CODEC.G.723.1--ITU-T specification of audio CODEC. G.726--ITU-T specification of audio CODEC.G.727--ITU-T specification of audio CODEC. G.728--ITU-T specification of audio CODEC.G.729--ITU-T specification of audio CODEC.Gatekeeper--A device that translates network addresses and aliases to make connections via the H.323 protocol on a packet-switched network.Gateway--A device that acts as an interface between two or more networks to connect dissimilar communications systems.Generic Traffic Shaping (GTS)--A mechanism to control the traffic flow on a particular interface.H.225.0--A protocol for call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats.H.235--For security in H.323 network.H.239--For dual stream use in videoconferencing.H.245--A control protocol for multimedia communication.H.246--ITU-T specification for H.323/PSTN Interworking.H.248--ITU-T standard for a centralized VOIP network. (Same as Megaco defined by IETF.)H.261--Used primarily in older videoconferencing and video telephony products.H.263--Used primarily for videoconferencing, video telephony, and internet video.H.264--Also known as MPEG-4 Part 10, or AVC (for Advanced Video Coding).H.323--An umbrella recommendation from the ITU-T that defines the protocols to provide audio-visual communication sessions on any packet network.H.350--Directory Services Architecture for Multimedia Conferencing.H.360--An architecture for end-to-end QoS control and signaling.H.450--For supplementary services such as call waiting, call forwarding, etc.H.460.x--Supplements in H.323.H.501--Protocol for mobility management and intra/inter-domain communication in multimedia systems.H.510--Mobility for H.323 multimedia systems.Hairpin--To send a call back in the direction that it came from.Hop off--Point at which a call transitions from H.323 to non-H.323, typically at a gateway.iLBC--Internet Low Bitrate Vocoder.Instant Messenging (IM)--A software that allows users to exchange messages in real time. For example, MSN Messenger, Yahoo! Messenger, etc.Internet telephony--Technologies and services of using the Internet for voice and multimedia communications.IntServ--An architecture which specifies the elements to guarantee quality of service (QoS) on networks.IP--Inernet Protocol.IP Centrex--Using IP-based network to provide centrex services such as call hold, call transfer, last number look-up and redial, call forward, three-way calling.IP fragmentation--IP datagrams to be fragmented into pieces small enough to pass over a link with a smaller MTU than the original datagram size.IP PBX--IP Private Branch Exchange. A telephone, data and video switching system, usually located on customer premises and belonging to the user.IP phone--A device that converts voice into digital packets and vice versa to make phone calls over Internet possible.IP telephony--Technologies and services for the two-way transmission of voice over IP network.IPDC--IP Device Control (protocol).IPDR--Internet Protocol Detail Record (protocol).ISDN--Integrated Services Digital Network.ITSP--Internet Telephony Service Provider.Jitter--A momentary fluctuation in the transmission signal.Lag--The extra time taken by a packet of data to travel from the source computer to the destination computer and back again.Latency--The time that elapses between the initiation of a request for data and the start of the actual data transfer.LDCELP--Low-Delay Code Excited Linear Prediction.LPC--Linear-Predictive Codec. LPCP--Lightweight Phone Control Protocol.MCML PPP--Multi-Class Multilink Point-to-Point Protocol.Media gateway (MG)--A translation unit between disparate telecommunications networks.Media gateway controller (MGC)--A system used in MGCP/H.248/Megaco VoIP telephony architectures to control a number of Media Gateways.Megaco--A IETF VOIP signaling protocol, same as H.248 of ITU-T.MGCP--Media Gateway Control Protocol.

Modulation--To carring information on a signal by varying one or more of the signal's basic characteristics -- frequency, amplitude and phase.MoIP--Modem over IP.MOS--Mean Opinion Score, a numerical indication of the perceived quality of received media after compression and/or transmission.MPEG-4 Part 10--Also known as H.264 or AVC, a video codec used for internet, broadcast, and on storage media.MPEG-4 Part 2--Used for internet, broadcast, and on storage media.MP-MLQ--Multi-Pulse, Multi-Level Quantization. MTU--Maximum Transmission Unit.NCS--Network-Based Call Signaling.Net Phone --A net phone uses the Voice over IP technology to make voice calls.Network convergence--The integration of all traffic types - voice, data and video solutions - onto a single IP network.NGN--Next Generation Network.OSP--Open Settlement Protocol.Packet loss--The loss of data packets during transmission over a computer network.Packet switched network--Networks that break messages into small packets, and route them across different channels to their destination where they are reassembled in their proper sequence.PBX--Private Branch Exchange is an in-house telephone switching system.PCM--Pulse Code Modulation.Peer-to-Peer (P2P)--A form of computing where two or more than two users can communicate directly without a central control point.Policy-based Routing--A technique used to make routing decisions based on policies set by the network administrator.POTS--Plain Old Telephone Service.PQ-CBWFQ--Priority Queuing - Class-Based Weighted Fair Queuing.PRI--Primary Rate Interface, a type of ISDN interface.Priority Queuing (PQ)--A queuing technique to give mission-critical traffic higher priority that less critical traffic.Processor drain--A drop in the quality of VoIP phone service when a user opens several applications on his computer simultaneously.Propogation delay--The time required for a signal to travel from one point to another.Proxy server-- Performs routing of a session invitations according to invitee's current location, authentication, accounting, etc.PSTN--Public Switched Telephone Network, refers to the telephone system that transmits analog voice data.Q.931--ISDN connection control protocol.QoS--Quality of Service.QSIG--Signaling standard for PBX.RADIUS--Remote Authenticaion Dial-In User Service.Random Early Detection (RED)--An active queue management algorithm. It is also a congestion avoidance algorithm.RAS--Registration, Admission, Status (RAS), a management protocol between terminals and Gatekeepers in the H.323 network.Redirect server--Receives a request and sends back a reply containing a list of the current location of a particular user.Registrar server-- Accepts REGISTER requests and places the information it receives in those requests into the location service for the domain in handles.RSVP-- Resource Reservation Protocol.RTCP-- Real Time Control Protocol.RTP-- Real Time Transport Protocol.RTSP--Real Time Streaming ProtocolRUDP--Reliable User Datagram Protocol.Sampling--A methodology used to measure the value of an analog signal at regular intervals, and encoding it into a digital format for phone services.Sampling rate--The number of samples per second taken from a continuous (analog) signal to make a discrete(digital) signal.SAP-- Session Announcement Protocol.SCCP--Skinny Client Control Protocol.SCTP--Stream Control Transmission Protocol.SDP--Session Description Protocol.Servie Level Agreement (SLA)--A contract between a network service provider and a customer that specifies what services and quality the service provider will furnish.Service provider-- A business entity that provides a communication, storage or processing service for a fee.SGCP-- Simple Gateway Control Protocol.Signaling gateway--A network component responsible for translating signaling messages between one medium (usually IP) and another (PSTN).SIGTRAN-- A family of protocols that provides reliable datagram service and user layer adaptations for SS7 and ISDN communications protocols.SIP--Session Initiation Protocol, an IP telephony signaling protocol.SIP phone-- A telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet.Skinny-- Skinny Client Control Protocol.Skype--A peer-to-peer Internet telephony company that leading the way voice calls are made by using VoIP technology.Soft switch-- A software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks.Softphone-- A software application that is installed in the user’s PC enables voice calls over the Internet.Softphone client-- The software installed in the user’s computer to make calls over the Internet.Speex--A free software speech codec.SRTP--Secure Real-time Transport Protocol.SS7--Signaling System number 7.T.120-- ITU-T specifcation for Real-time Conferencing.T.38-- ITU-T specification for Facsimile over IP.TAPI--Telephony API.ToS--Type of Service.Traffic shaping--To control network traffic in order to optimize or guarantee performance, low latency, and/or bandwidthUnified Messaging (UM)-- The integration of different streams of messages (e-mail, Fax, voice, video, etc.) into a single in-box, accessible from a variety of different devices.User Agents-- A software program installed in a user’s terminal or an IP phone to initiate and terminate phone calls.V.150-- ITU-T specification for Modem over IP.Voice chat-- An application that enables two or more individuals to carry on a verbal conversation (audio conference) over the Internet.Voice over IP (VOIP)--The technology that is used to transmit voice over the Internet.Voicemail-- A telephone messaging system that digitizes the analog voice signals and stores them on disk or flash memory in a central computer.VOIP Gateway-- A device provides the conversion interface between the PSTN and an IP network for voice and fax calls.VOIP PBX-- Voice over Internet Protocol Private Branch eXchange.VOIP Phone-- A device that uses the IP network to route voice calls by converting the voice data into IP packets and vice versa.VOIP services-- Services that use the IP network to move voice data.Web phone-- A device that allows users to make voice calls over the Internet.WFQ-- Weighted Fair Queuing, a packet scheduling technique allowing guaranteed bandwidth services. WiFi phone-- A device that enables users to make phone calls from WiFi network environments.WMV-- Windows Media Video. WRED--Weighted Random Early Drop/Detect.X264-- A GPL-licensed implementation of H.264 encoding standard.ZRTP-- An extension to RTP which describes a method of Diffie-Hellman key agreement for SRTP.

Network DictionaryNetwork Protocols Handbook Network Protocol Map Network Security Map Nerwork Management Map Wireless Technology Map

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