Grandstream Networks 200 Series user manual Key Features, Software Features

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3.1Key Features

Grandstream BudgeTone-200 IP Phone is a next generation IP telephone based on industry open standard SIP (Session Initiation Protocol). Built on innovative technology, Grandstream IP Phone features market leading superb sound quality and rich functionalities at mass-affordable price.

Software Features:

Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP/SNTP, TFTP.

Support multiparty conferencing

Support NAT traversal using IETF STUN and Symmetric RTP

Advanced Digital Signal Processing (DSP) technology to ensure superior hi- fidelity audio quality, interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server and Gateway products

Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology

Support popular codecs including G711 (a-law and u-law), G.723.1 (6.3K), G.729A/B and GSM. Dynamic negotiation of codec and voice payload length

Support standard voice features such as Caller ID Display or Block, Call Waiting, Call Waiting Caller ID, Call Hold, Call Transfer (attended/blind), Do-Not-Disturb, Call Forwarding, in-band and out-of-band DTMF(RFC2833), SIP INFO, Dial Plans, Off-Hook Auto Dial, Auto Answer, Early Dial and Speed Dial, etc.

Full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator, downloadable ring tone, etc.

Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168) and AGC (Automatic Gain Control)

Support Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for speakerphone mode

Support sidetone

Support DIGEST authentication and encryption using MD5 and MD5-sess

Provide easy configuration through manual operation (phone keypad), Web interface or automated provisioning by downloading encrypted configuration file via HTTP/TFTP for mass deployment

Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)

Support firmware upgrade via TFTP or HTTP.

Support DNS SRV Look up and SIP Server Fail Over

Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for speakerphone mode

Support for Authenticating configuration file before accepting changes

allow user to specify different URL for configuration file and firmware files

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Contents For Firmware Version Grandstream Networks, IncUsing BUDGETONE-200 IP Phone Table of ContentsRestore Factory Default Setting Appendix I Glossary of Terms Welcome Installation What is Included in the PackageConnecting Your Phone Warranty Safety CompliancesPage Product Overview Software Features Key FeaturesHardware Features Model BudgeTone-200 Hardware SpecificationUsing BudgeTone-200 IP Phone Getting Familiar with LCD / LEDIcon LCD Icon Definitions Page Getting Familiar with Keypad Called MenuCallers MessageMake Calls using Numbers Handset, Speakerphone and Headset ModeMaking and Answering Phone Calls SEND/REDIALAnswer an Incoming Call Make Calls using IP AddressMENUkey 192 168 000 MENUkey 192168001020*45062Call Hold Handset Mode, Speakerphone/Headset ModeCall Waiting and Call Flashing Call TransferConference Call Checking Message and Message Waiting IndicationAttended Transfer Mute and Delete Call FeaturesConfiguration Guide Configuration with KeypadMenu Item Menu Functions Display 7 G-711u Display 10 Phy Addr Configuration with Web Browser Access the Web Configuration MenuEnd User Configuration Status Basic Advanced Account Settings Grandstream Device ConfigurationEnd User Password IP Address Time Zone DaylightLAN Dhcp WAN MACDhcp IP DMZ IPDevice Mode PasswordWAN side http access End UserProduct Model BT200 PPPoE Link Up disabledCloned WAN MAC Addr LAN Dhcp Base IPAdvanced User Configuration Status Basic Account SettingsAutomatic Upgrade Timeout Use # as AdminSilence SuppressionLocal RTP UpgradePort Use RandomDtmf Update Disable CallOption 42 to Override NTPAuthenticate Password Authenticate IDAccount Active Account Name MyCompanyAuto Answer Account Name Individual Account SettingsSIP Server SIP User IDMWI Features Enable CallDisable Missed-Call TimerSpecial Feature Enable 100relUAS Specify Force InviteRebooting the Phone from Remote Saving the Configuration ChangesConfiguration through Central Provisioning Server Firmware Upgrade Upgrade through HttpUpgrade through Tftp Page Step Restore Factory Default SettingAppendix I Glossary of Terms Did DNSDSP FqdnFXO Echo CancellationFXS DhcpIP-PBX HttpIVR MTUNTP NATOBP/SBC PstnSIP SDPStun TCPVoIP Vlan