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Page 41 / 127 Scroll up to view Page 36 - 40
34
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
The Voice - SIP Screen
This screen lets you configure service, music-on-hold, group paging, call hunt, and auto-attendant settings.
SIP Parameters
Max Forward
. This is the SIP Max Forward value, which can range from 1 to 255. The default is
70
.
Max Redirection
. This is the number of times an invite can be redirected to avoid an infinite loop. The default
is
5
.
Max Auth
. This is the maximum number of times (from 0 to 255) a request may be challenged. The default is
2
.
SIP User Agent Name
. This is the User-Agent header used in outbound requests. The default is
$VERSION
.
SIP Server Name
. This is the Server header used in responses to inbound responses. The default is
$VERSION
.
SIP Reg User Agent Name
. This is the User-Agent name to be used in a REGISTER request. If this is not
specified, then the SIP User Agent Name will also be used for the REGISTER request.
SIP Accept Language
. This is the Accept-Language header used by the System. There is no default (this
indicates the System does not include this header).
DTMF Relay MIME Type
. This is the MIME Type used in a SIP INFO message to signal a DTMF event. The default
is
application/dtmf-relay
.
Hook Flash MIME Type
. This is the MIME Type used in a SIP INFO message to signal a hook flash event. The
default is
application/hook-flash
.
Remove Last Reg
. This feature lets you remove the last registration before registering a new one if the value is
different. Select
yes
or
no
from the drop-down menu. The default is
no
.
Use Compact Header
. This feature lets you use compact SIP headers in outbound SIP messages. Select
yes
or
no
from the drop-down menu. The default is
no
.
Escape Display Name
. This feature lets you keep the Display Name private. Select
yes
if you want the System to
enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any
Figure 6-14: Voice - SIP Screen - SIP Parameters
IMPORTANT:
In most cases, you should not change the service settings unless instructed to do
by your ITSP.
Page 42 / 127
35
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
occurrences of “ or \ in the string will be escaped with \” and \\ inside the pair of double quotes. Otherwise,
select
no
. The default is
no
.
SIP Timer Values (sec)
SIP T1
. This is the RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. The default is
.5
.
SIP T2
. This is the RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE
responses), which can range from 0 to 64 seconds. The default is
4
.
SIP T4
. This is the RFC 3261 T4 value (maximum duration a message will remain in the network), which can
range from 0 to 64 seconds. The default is
5
.
SIP Timer B
. This is the INVITE time-out value, which can range from 0 to 64 seconds. The default is
32
.
SIP Timer F
. This is the non-INVITE time-out value, which can range from 0 to 64 seconds. The default is
32
.
SIP Timer H
. This is the INVITE final response, time-out value, which can range from 0 to 64 seconds. The default
is
32
.
SIP Timer D
. This is the ACK hang-around time, which can range from 0 to 64 seconds. The default is
32
.
SIP Timer J
. This is the non-INVITE response, hang-around time, which can range from 0 to 64 seconds. The
default is
32
.
INVITE Expires
. This is the INVITE request Expires header value. If you enter 0, then the Expires header is not
included in the request. The default is
240
.
ReINVITE Expires
. This is the ReINVITE request Expires header value. If you enter 0, then the Expires header is
not included in the request. The default is
30
.
Reg Min Expires
. This is the minimum registration expiration time allowed from the proxy in the Expires header
or as a Contact header parameter. If the proxy returns a value less than this setting, then the minimum value is
used. The default is
1
.
Reg Max Expires
. This is the maximum registration expiration time allowed from the proxy in the Min-Expires
header. If the value is larger than this setting, then the maximum value is used. The default is
7200
.
Reg Retry Intvl
. This is the interval to wait before the System retries registration after failing during the last
registration. The default is
30
.
Figure 6-15: Voice - SIP Screen - SIP Timer Values
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36
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Reg Retry Long Intvl
. When registration fails with a SIP response code that does not match, the System will wait
for the specified length of time before retrying. If this interval is 0, then the System will stop trying. This value
should be much larger than the Reg Retry Intvl value. The default is
1200
.
Response Status Code Handling
SIT1-4 RSC
. Enter the SIP response status code for the appropriate SIT Tone (SIT stands for Special Information
Tone). For example, if you set the SIT1 RSC to 404, then when the user makes a call and a failure code of 404 is
returned, the SIT1 tone is played.
Try Backup RSC
. This is the SIP response code that retries a backup server for the current request.
Retry Reg RSC
. This is the interval to wait before the System retries registration after failing during the last
registration.
RTP Parameters
RTP Port Min
. This is the minimum port number for RTP transmission and reception. The default is
16384
.
RTP Port Max
. This is the maximum port number for RTP transmission and reception. The default is
16482
.
RTP Packet Size
. This is the packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a
multiple of 0.01 seconds. The default is
0.030
.
Max RTP ICMP Err
. This indicates that the RTP data stream has failed due to ICMP errors. The default is
0
.
RTCP Tx Interval
. This is the interval for sending out RTCP sender reports on an active connection. It can range
from 0 to 255 seconds. The default is
0
.
No UDP Checksum
. Select
yes
if you want the System to calculate UDP header checksum for SIP messages.
Otherwise, select
no
. The default is
no
.
Stats in BYE
. This sets whether the System will include the P-RTP-Stat header or response to a BYE message.
The header contains RTP statistics of the current call. Select
yes
or
no
from the drop-down menu. The default
is
no
.
SDP Payload Types
NSE Dynamic Payload
. This is the NSE dynamic payload type. The default is
100
.
AVT Dynamic Payload
. This is the AVT dynamic payload type. The default is
101
.
Figure 6-17: Voice - SIP Screen - RTP Parameters
Figure 6-16: Voice - SIP Screen - Response Status Code
Handling
Figure 6-18: Voice - SIP Screen - SDP Payload Types
Page 44 / 127
37
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
INFOREQ Dynamic Payload
. This is the INFOREQ dynamic payload type. There is no default.
G726r16 Dynamic Payload
. This is the G726-16 dynamic payload type. The default is
98
.
G726r24 Dynamic Payload
. This is the G726-24 dynamic payload type. The default is
97
.
G726r40 Dynamic Payload
. This is the G726-40 dynamic payload type. The default is
96
.
G729b Dynamic Payload
. This is the G729b dynamic payload type. The default is
99
.
NSE Codec Name
. This is the NSE codec name used in SDP. The default is
NSE
.
AVT Codec Name
. This is the AVT codec name used in SDP. The default is
telephone-event
.
G711u Codec Name
. This is the G711u codec name used in SDP. The default is
PCMU
.
G711a Codec Name
. This is the G711a codec name used in SDP. The default is
PCMA
.
G726r16 Codec Name
. This is the G726-16 codec name used in SDP. The default is
G726-16
.
G726r24 Codec Name
. This is the G726-24 codec name used in SDP. The default is
G726-24
.
G726r32 Codec Name
. This is the G726-32 codec name used in SDP. The is
G726-32
.
G726r40 Codec Name
. This is the G726-40 codec name used in SDP. The default is
G726-40
.
G729a Codec Name
. This is the G729a codec name used in SDP. The default is
G729a
.
G729b Codec Name
. This is the G729b codec name used in SDP. The default is
G729ab
.
G723 Codec Name
. This is the G723 codec name used in SDP. The default is
G723
.
NAT Support Parameters
Handle VIA received
. If you select yes, the System will process the received parameter in the VIA header (this is
inserted by the server in a response to any one of its requests). If you select no, the parameter will be ignored.
Select
yes
or
no
from the drop-down menu. The default is
no
.
Handle VIA rport
. If you select yes, the System will process the rport parameter in the VIA header (this is
inserted by the server in a response to any one of its requests). If you select no, the parameter will be ignored.
Select
yes
or
no
from the drop-down menu. The default is
no
.
Figure 6-19: Voice - SIP Screen - NAT Support
Parameters
Page 45 / 127
38
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Insert VIA received
. This lets you insert the received parameter into the VIA header of SIP responses if the
received-from IP and VIA sent-by IP values differ. Select
yes
or
no
from the drop-down menu. The default is
no
.
Insert VIA rport
. This feature lets you insert the rport parameter into the VIA header of SIP responses if the
received-from port and VIA sent-by port numbers differ. Select
yes
or
no
from the drop-down menu. The default
is
no
.
Substitute VIA Addr
. This feature lets you use NAT-mapped IP:port values in the VIA header. Select
yes
or
no
from the drop-down menu. The default is
no
.
Send Resp To Src Port
. This feature lets you send responses to the request source port instead of the VIA sent-
by port. Select
yes
or
no
from the drop-down menu. The default is
no
.
STUN Enable
. This feature lets you use STUN to discover NAT mapping. Select
yes
or
no
from the drop-down
menu. The default is
no
.
STUN Test Enable
. If the STUN Enable feature is enabled and a valid STUN server is available, then the System
can perform a NAT type discovery operation when it powers on. It will contact the configured STUN server, and
the result of the discovery will be reported in a Warning header in all subsequent REGISTER requests. If the
System detects symmetric NAT or a symmetric firewall, NAT mapping will be disabled.
The STUN Test Enable feature lets you use the STUN test. Select
yes
or
no
from the drop-down menu. The default
is
no
.
STUN Server
. Enter the IP address of the STUN server to contact for NAT mapping discovery.
EXT IP
. Enter the external IP address to substitute for the actual IP address of the System in all outgoing SIP
messages. If 0.0.0.0 is specified, then no IP address substitution will be performed.
EXT RTP Port Min
. This is the external port mapping number of the RTP Port Min. number. If this value is not
zero, then the RTP port number in all outgoing SIP messages will be substituted for the corresponding port value
in the external RTP port range.
NAT Keep Alive Intvl
. This is the interval between NAT-mapping, keep alive messages. The default is
15
.
PBX Parameters
Proxy Network Interface
. This tells the System how the clients (usually phones) are connected. Select
LAN
or
WAN
. The default is
WAN
.
Proxy Listen Port
. This is the port used by the System when it listens for client messages at the selected
interface. The default is
6060
.
Figure 6-20: Voice - SIP Screen - PBX Parameters

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