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Grandstream Networks BT200 User Manual

Page 34

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Grandstream Networks, Inc.

BT-200 User Manual

Page 34 of 35

Firmware 1.1.1.14

Last Updated: 12/2006

MTU A Maximum Transmission Unit (MTU) is the largest size

packet

or

frame

, specified in

octet

s (eight-

bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for
Ethernet is 1500 byte.

NAT
Network Address Translation

NTP
Network Time Protocol, a protocol to exchange and synchronize time over networks The port used
is UDP 123 Grandstream products using NTP to get time from Internet

OBP/SBC
Outbound Proxy or another name Session Border Controller. A device used in

VoIP

networks.

OBP/SBCs are put into the signaling and media path between calling and called Caller. The OBP/SBC
acts as if it was the called VoIP phone and places a second call to the called Caller. The effect of this
behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the
OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private
OBP/SBCs are used along with

firewalls

to enable VoIP calls to and from a protected enterprise network.

Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks
with

internet

connections using

NAT

.


PPPoE
Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.

PSTN
Public Switched Telephone Network. The phone service we use for every ordinary phone call, or
called POT (Plain Old Telephone), or circuit switched network.

RTCP
Real-time Transport Control Protocol, defined in

RFC 3550

, a sister protocol of the

Real-time

Transport Protocol

(RTP), It partners RTP in the delivery and packaging of multimedia data, but does not

transport any data itself. It is used periodically to transmit control packets to participants in a streaming
multimedia session. The primary function of RTCP is to provide feedback on the quality of service being
provided by RTP.

RTP
Real-time Transport Protocol defines a standardized packet format for delivering audio and video
over the Internet. It was developed by the Audio-Video Transport Working Group of the

IETF

and first

published in 1996 as

RFC 1889


SDP
Session Description Protocol is a format for describing

streaming media

initialization parameters. It

has been published by the

IETF

as

RFC

2327.


SIP
Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice
transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream
products are SIP based

STUN
Simple Traversal of UDP over NATs is a

network protocol

allowing clients behind

NAT

(or multiple

NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by
the NAT with a particular local port. This information is used to set up UDP communication between two
hosts that are both behind NAT routers. The protocol is defined in

RFC 3489

. STUN will usually work well

with non-symmetric NAT routers.

TCP Transmission Control Protocol is one of the core protocols of the

Internet protocol suite

. Using TCP,

applications on networked hosts can create connections to one another, over which they can exchange
data or

packets

. The protocol guarantees reliable and in-order delivery of sender to receiver data.


TFTP
Trivial File Transfer Protocol, is a very simple

file

transfer

protocol

, with the functionality of a very

basic form of

FTP

; It uses

UDP

(port 69) as its

transport protocol

.