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Using Session Initiated Protocol (SIP)
Using the SIP Phone
SIP Phone Features
Mute: When pressed, the SIP phone turns off the microphone signal
from the handset but will still play the voice from the other party.
Hold: When pressed, the SIP phone disconnects both the microphone
and the speaker while the connection is kept alive. No voice packets are
transmitted from the D-Link SIP phone. The hold LED is on. The user
may press the button again to release the call. This feature requires
support from the remote phone for proper functioning.
Redial: When pressed, the SIP phone redials last dialed number.
Transfer: Toggle the hook-switch quickly to flash the call. The SIP phone
will play a dial tone. Then dial the new party's number to transfer the call.
The SIP phone transfers the call and plays a busy tone. Flashing the
hook twice before dialing the number will restore the call to the normal
state (call active state).
Speaker: One-touch dialing key. When pressed, the speaker and speaker
LED are on while on-hook. If the user off-hooks after dialing or presses
the button again, one-touch operation is terminated and the LED and
speakers turn off.
Algorithms
Codecs: The D-Link SIP phone supports G.711 U/A law, G.723.1, and
G.729AB. The browser configuration allows selecting the codec and its
priority.
Voice Activity Detection, Silence Suppression, and Comfort Noise
Generation: The VAD can be disabled in the configuration irrespective
of the codec being used.
Adjustable Jitter Buffer: D-Link SIP phones uses a robust adaptive
jitter buffer algorithm. It can be disabled and a fixed size jitter buffer can
be used instead through configuration.
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