Grandstream Networks HT812 Series Administration Manual page 25

Conference URI Allows users to manually configure the conference URL. The default is null.
Use First
Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in
Matching
same order received in INVITE.
Vocoder in
Default is No.
200OK SDP
Preferred Configures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in SDP
Vocoder message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS
Voice Frames Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for
per TX G711/G726/G723/other codecs respectively.
Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps.
G723 Rate
Default is 6.3kbps.
iLBC Frame
Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms.
Size
Disable OPUS
Disables OPUS stereo in SDP. Default is No.
Stereo in SDP
Determines payload type for iLBC. Valid range is between 96 and 127.
iLBC Payload
Type Default is 97.
OPUS Payload
Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.
Type
Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of "silent packets"
over the network.
VAD
Default is No.
Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received
Symmetric RTP
by the device. Default is No.
Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through. If using Pass-through mode, select
Fax Mode
preference codec as PCMU or PCMA.
Re-Invite after
Fax Tone Permits the unit to send out the re-INVITE for T.38 or Fax Pass Through if a fax tone is detected. Default is Enabled
Detected
Jitter Buffer
Selects jitter buffer type (Fixed or Adaptive) based on network conditions.
Type
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.
Jitter Buffer
Medium (initial 100ms, min 20ms, max 200ms).
Length
Low (initial 50ms, min 10ms, max 100ms).
Selects SRTP mode to use ("Disabled", "Enabled but not forced", or "Enabled and forced"). Default is Disabled
It uses SDP Security Description to exchange key. Please refer to
SRTP Mode
SDES: https://tools.ietf.org/html/rfc4568
SRTP: https://www.ietf.org/rfc/rfc3711.txt
Allows users to select supported SRTP Key Length. the available values are :
1. AES 128&256 bit
SRTP Key
2. AES 128 bit
Length
3. AES 256 bit
Set to AES 128&256 bit By Default.
Crypto Life
Adds crypto life time header to SRTP packets. Default is Yes.
Time
SLIC Setting Depends on standard phone type (and location).
Caller ID
Selects the caller id scheme, for example: Bellcore/Telcordia, ETSI-FSK ...
Scheme
DTMF Caller ID Defines the start and stop tones (Default, A, B, C, D or #) to delimit CID.
Disable analog phone's caller ID when receiving a call with "Anonymous", "unavailable" or "unknown" in FROM
Disable
header and without "Display info".
Unknown Caller
Note: This relies also on analog phone's design, some phones will still display "unknown" with this feature enabled.
ID
Default is No.
Replace
Beginning '+' When this feature is set to Yes, device will replace the "+" sign at the beginning of a number in the FROM header.
with 00 in Caller Default is No.
ID
Polarity Reverses the polarity upon call establishment and termination.
Reversal
Default is No.
Loop Current Allows the traditional PBX used with HT81x to apply this method for signaling call termination. Method initiates
Disconnect short voltage drop on the line when remote (VoIP) side disconnects an active call. Default is No.