Settings/Sip/Codec - Grandstream Networks GVC3200 Administration Manual

It is used to define an estimate of the round trip time of transactions between a client
and server. If no response is received in T1, the figure will increase to 2*T1 and then
SIP T1 Timeout
4*T1. The request re-transmit retries would continue until a maximum amount of
time define by T2. The default setting is 0.5 second.
It is used to define the maximum retransmit time of any SIP request messages
SIP T2 Timeout (excluding the SIP INVITE message). The re-transmitting and doubling of T1
continues until it reaches the T2 value. The default setting is 4 second.
It is used to set if the device will remove outbound proxy URI from the Route header.
Remove OBP from This is used for the SIP Extension to notify the SIP server that the device is behind a
NAT/Firewall. If it is set to "Yes", it will remove the Route header from SIP requests.
Route
The default setting is "No".
Check Domain It is used to set if the device will check the domain certificates if TLS/TCP is used for
SIP Transport. The default setting is "No".
Certificates
It is used to configure the certificate for Authentication, and the option "Check
Domain Certificate
Domain certificates" needs to be set to "Yes".

SETTINGS/SIP/CODEC

Parameters Descriptions
It is used to set the parameter to specify the mechanism to transmit DTMF (Dual
Tone Multi-Frequency) signals. There are 3 supported modes: in audio, RFC2833,
or SIP INFO.


DTMF

The default setting is "RFC2833".
DTMF Payload
It is used to configure the RTP payload type that indicates the transmitted packet
contains DTMF digits. The valid range is from 96 to 127. The default setting is "101".
Type
It lists the available and enabled audio codecs for this account. Users can enable
the specific audio codecs by moving them to the Selected box and set them with a
Preferred Vocoder
priority order from top to bottom. This configuration will be included with the same
preference order in the SIP SDP message. The codec option includes "PCMU",
"PCMA", "Opus", "G.722", and "G.722.1".
Firmware Version 1.0.1.48
www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299
In audio
DTMF is combined in the audio signal (not very reliable with low-bit-rate
codecs).
RFC2833
Specify DTMF with RTP packet. Users could know the packet is DTMF in the
RTP header as well as the type of DTMF.
SIP INFO
Use SIP INFO to carry DTMF. The disadvantage of this mode is that it's easy to
cause desynchronized of DTMF and media packet if the SIP and RTP
messages are required to transmitted respectively.
GVC3200/GVC3202 Administration Guide Page 42 of 71