Grandstream GXW410x User Manual - Manual (Page 17)

Grandstream Networks, Inc. GXW410x User Manual Page 17 of 32
Firmware Version 1.4.1.5 Last Updated: 5/2014
used payload type negotiated between the 2 conversation parties at run time.
e.g., if the first vocoder is configured as G723 and the “Voice Frames per TX” is set to
be 2, then th
e “ptime” value in the SDP message of an INVITE request will be 60ms
because each G723 voice frame contains 30ms of audio. Similarly, if this field is set
to be 2 and if the first vocoder chosen is G729 or G711 or G726, then the “ptime”
value in the SDP message of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the
BudgeTone 200 will use and save the maximum allowed value for the corresponding
first vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726, it
is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms)
and 64 (x2.5ms) frames respectively.
Local RTP port This parameter defines the local RTP-RTCP port pair the GXW410x will listen and
transmit. It is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use
port_value+2 for RTP and port_value+3 for its RTCP and so on. The default value is
5004.
RTP Loopback Default value is No. If set to Yes, means no RTP if RTP streams between 2 internal
ports.
Channel Settings
DTMF Method This parameter specifies the mechanism to transmit DTMF digits. There are7 modes
supported: in audio which means DTMF is combined in audio signal (not very reliable
with low bit-rate codec), via RTP (RFC2833), or via SIP INFO. Multiple DTMF
transmission schemas can be selected.

1 – in-audio

2 – RFC2833

3 – in-audio and RFC2833

4 – SIP Info

5 – in-audio and RFC2833

6 – SIP Info and RFC2833
1. 7 – in-audio, RFC2833, and SIP Info
No Key Entry Timeout Default is 4 seconds.
Local SIP Listen Port Default is ch1-8:5060++;. The ++ indicates increments by 2, so port 1 is set at 5060,
port at 5062 and so on. This setting can be used with Round Robin and/or Flexible
setting below to configure different ports to be placed under different Round Robin
groups.
SRTP Mode Default is disabled for all ports. The user can select to either enable it but not force it
or force it on an individual port basis. When used the communication will be sent
using Secure RTP.
Unconditional Call
Forward to VOIP:
This is an extremely important setting to make sure incoming PSTN calls are picked
up and forwarded to the correct VOIP destination.
User ID - This parameter allows users to configure a User ID or extension number to
be automatically dialed upon FXO line off-hook.
SIP Server - You also need to specify the Profile of the user id configured above (p1
stands for Profile 1, p2 stands for Profile 2 and so on).
SIP Destination Port - Along with the user-id and Profile, you also have the option to
choose the destination port where you would like to send the call. By default it should
be set to ch1-x:5060; (x can be 4 or 8 depending on number of ports).
We can also specify a different destination for each port. For example under User ID
we can type in: ch1:104;ch2:227;ch3-5:501;ch6,7:856.
Under Sip Server we can type in: ch1:p1;ch2-4:p2;ch5:p3
Under Sip Destination Port we can type in: ch1-2:5060;ch2:7080;ch3-8:5066++
T.38 Setting This setting allows you to make several options related to facsimile.
You can select the method: T.38 or Pass through (G711)
You can select the fax transmission rates (2400/4800/7200/9600/12000/14400bps)
You can enable or disable ECM (Error Checking Mode)
Note:
The user can only test the parameters for only one of the PSTN lines at the
same time. In all cases please enter the telephone numbers as if the lines were to