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Grandstream Networks, Inc. HT-386 User Manual Page 33 of 34
Firmware 1.0.3.64 Last Updated: 2/2007
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and
perhaps other media.
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 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.