Description:
Voice over IP (VOIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network. Using VOIP protocols, voice communications can be achieved on any IP network regardless it is Internet, Intranets or Local Area Networks (LAN). In a VOIP enabled network, the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. VOIP signaling protocols are used to set up and tear down calls, carry information required to locate users and negotiate capabilities.
Contents:
H.225: Call Signaling and RAS in H.323 VOIP Architecture
H.225.0, a key protocol in the H.323 VOIP architecture defined by ITU-T,
is a standard to cover narrow-band visual telephone services defined in
H.200/AV.120-Series Recommendations.
H.235: Security for H.323 based systems and communications
| H.235 is the security recommendation for the H.3xx series systems. In
particular, H.235 provides security procedures for H.323, H.225.0, H.245
and H.460 based systems. |
H.245: Control Protocol for Multimedia Communication
| H.245, a control signaling protocol in the H.323 multimedia
communication architecture, is for of the exchange of end-to-end H.245
messages between communicating H.323
endpoints/terminals. |
H.323: Packet-based multimedia communications (VoIP) architecture
H.323, a protocol suite defined by ITU-T, is for voice transmission over
internet (Voice over IP or VOIP). In addition to voice applications, H.323
provides mechanisms for video communication and data collaboration, in
combination with the ITU-T T.120 series standards. H.323 is one o the major VOIP
standards, just as Megaco and SIP.
Megaco / H.248: Media Gateway Control protocol
| Megaco/H.248, the Media Gateway Control Protocol, is for control of
elements in a physically decomposed multimedia gateway, which enables
separation of call control from media
conversion. |
MGCP: Media Gateway Control Protocol
| Media Gateway Control Protocol (MGCP) is a VOIP protocol used between
elements of a decomposed multimedia gateway which consists of a Call
Agent, which contains the call control "intelligence", and a media gateway
which contains the media functions, e.g., conversion from TDM voice to
Voice over IP. |
NCS: Network-Based Call Signaling Protocol
|
Network-based Call Signaling, based on the Media Gateway Control
Protocol (MGCP), is the VOIP signaling protocol adobted by the CableLab as
a standard for PacketCable embbed clients, which is a network element that
provides |
SAP: Session Announcement Protocol
| Session Announcement Protocol (SAP) is an announcement protocol that
is used to assist the advertisement of multicast multimedia conferences
and other multicast sessions, and to communicate the relevant session
setup information to prospective participants.
|
SCCP: Skinny Client Control Protocol
Signaling Connection Control Part (SCCP), a routing protocol in SS7
protocol suite in layer 4, provides end-to-end routing for TCAP messages to
their proper database.
SDP: Session Description Protocol
The Session Description Protocol (SDP) describes multimedia sessions for
the purpose of session announcement, session invitation and other forms of
multimedia session initiation.
SIP: Session Initiation Protocol
| SIP (Session Initiation Protocol) is an application-layer control
protocol that can establish, modify, and terminate multimedia sessions
such as Internet telephony calls (VOIP). |
T.120: Multipoint Data Conferencing Protocol Suite
| The ITU T.120 standard is made up of a suite of communication and
application protocols, which are designed for multipoint Data Conferencing
and real time communication including multilayer protocols which
considerably enhance multimedia, MCU and codec control
capabilities. |
RTCP: RTP Control Protocol
| The Real Time Transport Control Protocol (RTP control protocol or
RTCP) is based on the periodic transmission of control packets to all
participants in the session, using the same distribution mechanism as the
data packets. |
RTP: Real Time Transport Protocol
The Real-Time Transport protocol (RTP) provides end-to-end delivery
services for data with real-time characteristics, such as interactive audio and
video or simulation data, over multicast or unicast network
services.
G.7xx: Audio (Voice) Compression Protocols (G.711, G.721, G.722, G.723, G.726, G.727. G.728, G.729)
G.7xx, including G.711, G.721, G.722, G.726, G.727, G.728, G.729, is an
suite of ITU-T standards for audio compression and de-commpression. It is
primarily used in telephony. In telephony, there are 2 main algorithms defined
in the standard.
H.261: Video Coding and Decoding (CODEC)
| H.261 is video coding standard by the ITU. It was designed for data
rates which are multiples of 64Kbit/s, and is sometimes called p x
64Kbit/s (p is in the range 1-30). |
H.263: Video Coding and Decoding (CODEC)rk
The H.263, by the International Telecommunications Union (ITU), supports
video compression (coding) for video-conferencing and video-telephony
applications.
H.264/MPEG-4: Video CODEC For High Quality Video Streaming
The H.264 and the MPEG-4 Part 10, also named Advanced Video Coding (AVC),
is jointly developed by ITU and ISO. H.264/MPEG-4 supports video compression
(coding) for video-conferencing and video-telephony applications.
COPS: Common Open Policy Service
The Common Open Policy Service (COPS) protocol is a simple query and
response protocol that can be used to exchange policy information between a
policy server (Policy Decision Point or PDP) and its clients (Policy Enforcement
Points or PEPs).
RTSP: Real Time Streaming Protocol
The Real-Time Streaming Protocol (RTSP) establishes and controls either a
single or several time-synchronized streams of continuous media such as audio
and video. RTSP does not typically deliver the continuous streams itself,
although interleaving of the continuous media stream with the control stream is
possible.
SCTP: Stream Control Transmission Protocol
Stream Control Transmission Protocol (SCTP) is designed to transport PSTN
signalling messages over IP networks, but is capable of broader
applications.
SIGTRAN: Signaling Transport Protocol Stack
|
Signaling Transport (SIGTRAN) refers to a protocol stack for the
transport of Switched Circuit Network (SCN) signaling protocols (such as
SS7/C7 an Q.931) over an IP network. |
TRIP: Telephony Routing Over IP
| Telephony Routing over IP (TRIP) is a policy driven
inter-administrative domain protocol for advertising the reachability of
telephony destinations between location servers, and for advertising
attributes of the routes to those
destinations. |
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