P a g e
|
44
GVC3210 Administration Guide
Version 1.0.1.21
SIP T1 Timeout
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
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.
SIP T2 Timeout
It is used to define the maximum retransmit time of any SIP request messages
(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.
Remove OBP
from Route
It is used to set if the device will remove outbound proxy URI from the Route header.
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.
The default setting is
“No”.
Check Domain
Certificates
It is used to set if the device will check the domain certificates if TLS/TCP is used for
SIP Transport.
Validate
Certification
Chain
When the SIP transport protocol is “TLS” and this option is enabled, the certificates in
device system and the trusted CA certificates uploaded by the user will be validated.
Settings/SIP/Codec
Parameters
Descriptions
Audio
DTMF
It is used to set the parameter to specify the mechanism to transmit DTMF (Dual Tone
Multi-Frequency) signals. Default setting is
“RFC2833”. There are 3 supported modes:
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 is easy to
cause desynchronized of DTMF and media packet if the SIP and RTP messages
are required to transmitted respectively.
DTMF Payload
Type
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”.
Preferred
Vocoder
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 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”, “G.722.1”, “G.722.1c”, “iLBC”, “G.729AB”.
