When it comes to VoIP, everyone wants a great phone to compliment his
or her system. Some phones come free when you purchase your
service, others you must pay for additionally. Here are the top
Six VoIP phones.
Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
Support NAT traversal via STUN & symmetric RTP
Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B,
G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model
102D). Dynamic negotiation of codec and voice payload length
Support standard voice features such as Caller ID Display or
Block, Call Waiting, Hold, TransfForward, FLASH, in-band and
out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial,
configurable emergency dialing (e.g., 911), early dial, click-to-dial
Support 3-way conferencing (Model 102D), full duplex
hands-fredomain acoustic echo cancellation (pending), redial, call log,
volume control, voice mail with indicator, downloadable ring tone
(pending)
Support Silence Suppression, VAD (Voice Activity Detection), CNG
(Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC
(Automatic Gain Control)
Support DIGEST authentication and encryption using MD5 and MD5-sess.
Provide easy configuration thru manual operation (phone keypad
and Web interface) or personalized automated provisioning via central
configuration file for mass deployment.
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
Support for fail-over SIP server and DNS server (pending)
Grandstream BudgeTone 102
Features Include:
Two RJ45 10Base-T Ethernet ports
Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
Support NAT traversal via STUN & symmetric RTP
Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B,
G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model
102D). Dynamic negotiation of codec and voice payload length
Support standard voice features such as Caller ID Display or
Block, Call Waiting, Hold, TransfForward, FLASH, in-band and
out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial,
configurable emergency dialing (e.g., 911), early dial, click-to-dial
Support 3-way conferencing (Model 102D), full duplex
hands-fredomain acoustic echo cancellation (pending), redial, call log,
volume control, voice mail with indicator, downloadable ring tone
(pending)
Support Silence Suppression, VAD (Voice Activity Detection), CNG
(Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC
(Automatic Gain Control)
Support DIGEST authentication and encryption using MD5 and MD5-sess.
Provide easy configuration thru manual operation (phone keypad
and Web interface) or personalized automated provisioning via central
configuration file for mass deployment.
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
NAT-friendly remote software upgrades capability (via tftp) even from behind firewalls/NATs.
Support for fail-over SIP server and DNS server (pending)