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Page 206 / 286 Scroll up to view Page 201 - 205
SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
204
B
Voice tab > SIP page
SIP Timer Values (sec) section
RFC 2543 Call Hold
If set to yes, unit will include c=0.0.0.0 syntax in SDP when
sending a SIP re-INVITE to the peer to hold the call. If set to
no, unit will not include the c=0.0.0.0 syntax in the SDP. The
unit will always include a=sendonly syntax in the SDP in
either case.
Default:
yes
Mark All AVT Packets
If set to yes, all AVT tone packets (encoded for redundancy)
have the marker bit set. If set to no, only the first packet has
the marker bit set for each DTMF event.
Default:
yes
SIP TCP Port Min
The lowest TCP port number that can be used for SIP
sessions.
Default:
5060
SIP TCP Port Max
The highest TCP port number that can be used for SIP
sessions.
Default:
5080
SIP T1
RFC 3261 T1 value (RTT estimate), which can range from 0 to
64 seconds.
Default:
.5
SIP T2
RFC 3261 T2 value (maximum retransmit interval for non-
INVITE requests and INVITE responses), which can range
from 0 to 64 seconds.
Default:
4
SIP T4
RFC 3261 T4 value (maximum duration a message remains in
the network), which can range from 0 to 64 seconds.
Default:
5
SIP Timer B
RFC 3261 INVITE transaction time-out value, which can range
from 0 to 64 seconds.
Default:
32
SIP Timer F
RFC 3261 Non-INVITE transaction
time-out value, which can
range from 0 to 64 seconds.
Default:
32
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SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
205
B
SIP Timer H
RFC 3261 time-out value for ACK receipt, which can range
from 0 to 64 seconds.
Default:
32
SIP Timer D
RFC 3261 wait time for response retransmits, which can
range from 0 to 64 seconds.
Default:
32
SIP Timer J
RFC 3261 wait time for Non-INVITE response hang-around
time, which can range from 0 to 64 seconds.
Default:
32
INVITE Expires
INVITE request Expires header value. If you enter 0, the
Expires header is not included in the request. Range: 0–(2
31
–
1).
Default:
240
ReINVITE Expires
ReINVITE request Expires header value. If you enter 0, the
Expires header is not included in the request. Range: 0–(2
31
–
1).
Default:
30
Reg Min Expires
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, the minimum
value is used.
Default:
1
Reg Max Expires
Maximum registration expiration time allowed from the proxy
in the Min-Expires header. If the value is larger than this
setting, the maximum value is used.
Default:
7200
Reg Retry Intvl
Interval to wait before the SPA9000 retries registration after
failing during the last registration.
Default:
30
Reg Retry Long Intvl
When registration fails with a SIP response code that does
not match<Retry Reg RSC>, the SPA9000 waits for the
specified length of time before retrying. If this interval is 0, the
SPA9000 stops trying. This value should be much larger than
the Reg Retry Intvl value, which should not be 0.
Default:
1200
Reg Retry Random Delay
Random delay range (in seconds) to add to <Register Retry
Intvl> when retrying REGISTER after a failure.
Default:
0
(disabled)
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SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
206
B
Voice tab > SIP page
Response Status Code Handling section
The RSC handling defines the behavior of the user audio tones played under
specific conditions signaled by the network, such as congestion, queuing, etc. The
default settings (blank) are adequate is most circumstances.
NOTE
These settings need to be compatible with the ITSP network settings. The ITSP
you use will inform you about any specific setting you need to modify on these
areas. These parameters have impact on the signaling and audio reliability.
Reg Retry Long Random
Delay
Random delay range (in seconds) to add to <Register Retry
Long Intvl> when retrying REGSITER after a failure.
Default:
0
(disabled)
Reg Retry Intvl Cap
The maximum value to cap the exponential back-off retry
delay (which starts at <Register Retry Intvl> and doubles on
every REGISTER retry after a failure). In other words, the retry
interval is always at <Register Retry Intvl> seconds after a
failure. If this feature is enabled, <Reg Retry Random Delay>
is added on top of the exponential back-off adjusted delay
value.
Default:
0
(disables the exponential back-off feature)
SIT1 RSC
SIP response status code for the appropriate Special
Information Tone (SIT). For example, if you set the SIT1 RSC
to 404, when the user makes a call and a failure code of 404
is returned, the SIT1 tone is played. Reorder or Busy Tone is
played by default for all unsuccessful response status code
for SIT 1 RSC through SIT 4 RSC.
SIT2 RSC
SIP response status code to INVITE on which to play the SIT2
Tone.
SIT3 RSC
SIP response status code to INVITE on which to play the SIT3
Tone.
SIT4 RSC
SIP response status code to INVITE on which to play the SIT4
Tone.
Try Backup RSC
SIP response code that retries a backup server for the
current request.
Retry Reg RSC
Interval to wait before the SPA9000 retries registration after
failing during the last registration.
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SPA9000 Field Reference
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SPA9000 Voice System Administration Guide
207
B
Voice tab > SIP page
RTP Parameters section
The RTP parameters define the specification of the RTP audio packets. The most
important parameter is the RTP Packet size (time) which defines the interval of
transmission of the RTP packets. It is extremely important that time interval
matches the ITSP settings. Its default value is 0.030 (30 milliseconds).
NOTE
These settings need to be compatible with the ITSP network settings. The ITSP
you use will inform you about any specific setting you need to modify on these
areas. These parameters have impact on the signaling and audio reliability.
RTP Port Min
Minimum port number for RTP transmission and reception.
<RTP Port Min> and <RTP Port Max> should define a range
that contains at least 4 even number ports, such as 100 –
106.
Default:
16384
RTP Port Max
Maximum port number for RTP transmission and reception.
Default:
16482
RTP Packet Size
Packet size in seconds, which can range from 0.01 to 0.16.
Valid values must be a multiple of 0.01 seconds.
Default:
0.030
Max RTP ICMP Err
Number of successive ICMP errors allowed when
transmitting RTP packets to the peer before the SPA9000
terminates the call. If value is set to 0, the SPA9000 ignores
the limit on ICMP errors.
Default:
0
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Page 210 / 286
SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
208
B
Voice tab > SIP page
SDP Payload Types section
The SDP Payload types defines the naming/numbering conventions for the audio
codecs used by the SPA9000 when communicating with the ITSP network.
Naming should match the ITSP names used. The default values are adequate in
most circumstances.
These settings need to be compatible with the ITSP network settings. The ITSP
you use will inform you about any specific setting you need to modify on these
areas. These parameters have impact on the signaling and audio reliability.
RTCP Tx Interval
Interval for sending out RTCP sender reports on an active
connection. It can range from 0 to 255 seconds. During an
active connection, the SPA9000 can be programmed to send
out compound RTCP packet on the connection. Each
compound RTP packet except the last one contains a SR
(Sender Report) and a SDES.(Source Description). The last
RTCP packet contains an additional BYE packet. Each SR
except the last one contains exactly 1 RR (Receiver Report);
the last SR carries no RR. The SDES contains CNAME, NAME,
and TOOL identifiers. The CNAME is set to <User
ID>@<Proxy>, NAME is set to <Display Name> (or
Anonymous if user blocks caller ID), and TOOL is set to the
Vendor/Hardware-platform-software-version (such as
Linksys/SPA9000-1.0.31(b)). The NTP timestamp used in the
SR is a snapshot of the SPA9000’s local time, not the time
reported by an NTP server. If the SPA9000 receives a RR
from the peer, it attempts to compute the round trip delay and
show it as the <Call Round Trip Delay> value (ms) in the Info
section of SPA9000 web page.
Default:
0
No UDP Checksum
Select yes if you want the SPA9000 to calculate the UDP
header checksum for SIP messages. Otherwise, select no.
Default:
no
Stats In BYE
Determines whether the SPA9000 includes the P-RTP-Stat
header or response to a BYE message. The header contains
the RTP statistics of the current call. Select yes or
no from the
drop-down menu. The format of the P-RTP-Stat header is:
P-RTP-State: PS=<packets sent>,OS=<octets
sent>,PR=<packets received>,OR=<octets
received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in
ms>,DU=<call duration in s>,EN=<encoder>,DE=<decoder>.
Default:
no
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