D-Link DPH-80 Manual - Page 54
SIP Phone Features, Algorithms, Redial, Transfer, Speaker, Codecs, Voice Activity Detection
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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. 54
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