Grandstream Networks, Inc.
HT-502 User Manual
Page 30 of 31
Firmware 1.0.0.29
Last Updated: 01/2007
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
.
