Grandstream Networks GXP – 2000 User Manual page 32

Sip enterprise phone
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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-
bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximu
Ethernet is 1500 byte.
NAT Network Address Translation
NTP Network Time Protocol, a protocol to e
is UDP 123 Grandstream products using NTP to get time from Internet
O
BP/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 party. The OBP/SBC
acts as if it was the called VoIP phone
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 t
OBP/SBCs are used along with firewalls to enable VoIP calls to and from a prot
P
ublic 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, o
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
transport any data itself. It is used periodically to trans
m
ultimedia 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 wa
s developed by the Audio-Video Transport Working Group of the IETF and first
p
ublished in 1996 as RFC 1889
S
DP 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 Grandst
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 we
with non-symmetric NAT routers.
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP
a
pplications 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 Trans
fer Protocol, is a very simple file transfer protocol, with the functionality of a very
b
asic form of FTP; It uses UDP (port 69) as its transport protocol.
Grandstream Networks, Inc.
xchange and synchronize time over networks The port used
and places a second call to the called party. The effect of this
ravels directly between the VoIP phones. Private
mit control packets to participants in a streaming
.
GXP-2000 Users Manual
Firmware 1.1.1.14
m for
ected enterprise network.
not
ream
Page 32 of 33
Last Updated: 12/2006
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