Grandstream Networks BT200 user manual NAT Network Address Translation

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NAT Network Address Translation

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.

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.

Grandstream Networks, Inc.

BT200 User Manual

Page 36 of 37

 

Firmware 1.1.1.14

Last Updated: 12/2006

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Contents BT200 User Manual Grandstream Networks, IncBT-Series BT200 Dual Ethernet Port SIP Phone Firmware VersionTABLE OF CONTENTS BT200 USER MANUALAccess the Web Configuration Menu Placing a Quick IP CallTABLE OF FIGURES TABLE OF TABLESGUI INTERFACES SAFETY COMPLIANCES WARRANTYWELCOME CONNECTING YOUR PHONE EQUIPMENT PACKAGINGCONNECT YOUR PHONE CONNECT YOUR PHONE TO THE VOIP NETWORK KEY FEATURES HARDWARE FEATURESPRODUCT OVERVIEW Lines Protocol Support1 Display Feature Keys Ethernet Ports Device Management Audio Features Call Handling FeaturesNetwork and Provisioning Advanced Server Features Security TABLE 3 TECHNICAL SOFTWARE FEATURESUSING THE BT200 IP PHONE GETTING FAMILIAR WITH THE LCD / LEDAlarm Clock Icon IconGETTING FAMILIAR WITH KEYPAD Menu and Call Keys Advanced Feature Buttons Speaker, Send Mute/DelAM PM Real-time ClockKey Button Key Button DefinitionsUP ↑ DOWN ↓PLACING CALLS USING AN IP ADDRESS MAKING AND ANSWERING PHONE CALLSHANDSET, SPEAKERPHONE AND HEADSET MODE PLACING A TRADITIONAL PHONE CALLPlacing a Quick IP Call ANSWER AN INCOMING CALLCall Hold Call Waiting and Call FlashingBlind and Attended Call Transfer Expected outcomes3-way Conference Retrieving MessagesCALL FEATURES Call FeaturesDisable Call Waiting per Call Enable Call Waiting per CallCONFIGURATION GUIDE CONFIGURATION WITH KEYPADMenu Item Menu FunctionsCONFIGURATION WITH WEB BROWSER Access the Web Configuration MenuEND USER CONFIGURATION End User Password IP Address Time Zone Daylight Savings Time ExamplesWAN side HTTP access Reply to ICMP on WAN port Cloned WAN MAC Addr Date Display Format Display Account Name instead of Date Device ModeLAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port ForwardingADVANCED USER CONFIGURATION MAC AddressIP Address Product ModelAdmin Password Silence Suppression Voice Frames per TX Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use # as Send Key Local RTP portFirmware Upgrade and provisioning Via TFTP Server Via HTTP Server Allow DHCP Option 66 to override server Automatic UpgradeAuthenticate Conf File Off-hook Auto Dial Use Random Port Keep-alive interval Use NAT IP STUN ServerDisable Call Waiting DTMF Payload Type Syslog Server Syslog Level NTP serverAllow DHCP Option 42 to override NTP server Distinctive Ring Tone Lock keypad update Authenticate IDAuthenticate PasswordSIP Registration Unregister on Reboot Local SIP port SIP T1 Timeout SIP T2 Interval NAT TraversalEarly Dial Dial Plan Prefix Enable Call Features Disable Missed-Call Register ExpirationSession Expiration Min-SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITEEnable 100rel Account Ring Tone Send Anonymous Auto Answer Allow Auto Answer by Call-Info Turn off speaker on remote disconnectSAVING THE CONFIGURATION CHANGES Special FeatureREBOOTING THE PHONE FROM REMOTE Check SIP User ID for Incoming Invite Preferred Vocoder SRTP ModeCONFIGURATION THROUGH CENTRAL PROVISIONING SERVER FIRMWARE UPGRADE UPGRADE THROUGH HTTPUPGRADE THROUGH TFTP Keypad Configuration Prompt MenuNOTE for TFTP/HTTP Firmware Upgrade No Local TFTP ServerRESTORE FACTORY DEFAULT SETTING Directions for RestorationGLOSSARY OF TERMS FXS is complimentary to FXS and the PSTN NAT Network Address Translation UDP User Datagram Protocol UDP is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages known as datagrams to one another. UDP does not provide the reliability and ordering guarantees that TCP does datagrams may arrive out of order or go missing without notice. However, as a result, UDP is faster and more efficient for many lightweight or time-sensitive purposes