Grandstream Networks 200 User Manual

 

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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. 
 

•  Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, HTTP, ARP/RARP, ICMP, 

DNS, DHCP, NTP/SNTP, TFTP. 

•  Support multiparty conferencing  

•  Supports Quick IP Call Mode. 
•  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 3

rd

 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