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GRANDSTREAM HT812 Analog FXS IP Gateway 2 Port + NAT Router User Guide

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  
Vocoder in  
200OK SDP  
same order received in INVITE.  
Default is No.  
Preferred  
Vocoder  
Configures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in SDP  
message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS  
Voice Frames  
per TX  
Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for  
G711/G726/G723/other codecs respectively.  
Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps.  
Default is 6.3kbps.  
G723 Rate  
iLBC Frame  
Size  
Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms.  
Disables OPUS stereo in SDP. Default is No.  
Disable OPUS  
Stereo in SDP  
Determines payload type for iLBC. Valid range is between 96 and 127.  
Default is 97.  
iLBC Payload  
Type  
OPUS Payload  
Type  
Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.  
Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets”  
over the network.  
Default is No.  
VAD  
Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received  
by the device. Default is No.  
Symmetric RTP  
Fax Mode  
Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through. If using Pass-through mode, select  
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  
Selects jitter buffer type (Fixed or Adaptive) based on network conditions.  
Detected  
Jitter Buffer  
Type  
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.  
Medium (initial 100ms, min 20ms, max 200ms).  
Jitter Buffer  
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  
Allows users to select supported SRTP Key Length. the available values are :  
1. AES 128&256 bit  
2. AES 128 bit  
SRTP Key  
Length  
3. AES 256 bit  
Set to AES 128&256 bit By Default.  
Crypto Life  
Time  
Adds crypto life time header to SRTP packets. Default is Yes.  
Depends on standard phone type (and location).  
SLIC Setting  
Caller ID  
Scheme  
Selects the caller id scheme, for example: Bellcore/Telcordia, ETSI-FSK …  
Defines the start and stop tones (Default, A, B, C, D or #) to delimit CID.  
DTMF Caller ID  
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  
ID  
Note: This relies also on analog phone’s design, some phones will still display “unknown” with this feature enabled.  
Default is No.  
Replace  
Beginning ‘+’  
with 00 in Caller  
ID  
When this feature is set to Yes, device will replace the “+” sign at the beginning of a number in the FROM header.  
Default is No.  
Reverses the polarity upon call establishment and termination.  
Default is No.  
Polarity  
Reversal  
Loop Current  
Disconnect  
Allows the traditional PBX used with HT81x to apply this method for signaling call termination. Method initiates  
short voltage drop on the line when remote (VoIP) side disconnects an active call. Default is No.  
GeneralDetails
NameGRANDSTREAM HT812 Analog FXS IP Gateway 2 Port + NAT Router User Guide
MakeGRANDSTREAM
LanguageEnglish
FiletypePDF (Download)
File size MB
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GRANDSTREAM HT812 Analog FXS IP Gateway 2 Port + NAT Router User Guide