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Page 96 / 186 Scroll up to view Page 91 - 95
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
SIP Tab
SIP Timer Values (sec)
Use Compact Header
Lets you use compact SIP headers in outbound SIP messages. Select yes
or
no from the
drop-down menu. If set to yes, the Linksys IP phone uses compact SIP headers in
outbound SIP messages. If set to no, the Linksys IP phone uses normal SIP headers. If
inbound SIP requests contain compact headers, Linksys IP phone reuses the same compact
headers when generating the response regardless the settings of the <Use Compact
Header> parameter. If inbound SIP requests contain normal headers, Linksys IP phone
substitutes those headers with compact headers (if defined by RFC 261) if <Use Compact
Header> parameter is set to yes.
The default is no.
Escape Display Name
Lets you keep the Display Name private. Select yes if you want the Linksys IP phone to
enclose the string (configured in the Display Name) in a pair of double quotes for
outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside
the pair of double quotes. Otherwise, select no.
The default is no.
RFC 2543 Call Hold
The default is no.
Mark All AVT Packets
SIP TCP Port Min
SIP TCP Port Min
Specifies the lowest TCP port number that can be used for SIP sessions.
SIP TCP Port Max
Specifies the highest TCP port number that can be used for SIP sessions.
CTI Enable
Enables or disables the Computer Telephone Interface feature provided by some servers.
Field
Description
SIP T1
RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.
The default is.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.
The default is 4.
SIP T4
RFC 3261 T4 value (maximum duration a message remains in the network), which can
range from 0 to 64 seconds.
The default is 5.
SIP Timer B
INVITE time-out value, which can range from 0 to 64 seconds.
The default is 32.
SIP Timer F
Non-INVITE time-out value, which can range from 0 to 64 seconds.
The default is 32.
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Linksys ATA Field Reference
SIP Tab
SIP Timer H
INVITE final response, time-out value, which can range from 0 to 64 seconds.
The default is 32.
SIP Timer D
ACK hang-around time, which can range from 0 to 64 seconds.
The default is 32.
SIP Timer J
Non-INVITE response hang-around time, which can range from 0 to 64 seconds.
The default is 32.
INVITE Expires
INVITE request Expires header value. If you enter 0, the Expires header is not included
in the request.
The default is 240. Range: 0–(2
31
–1).
ReINVITE Expires
ReINVITE request Expires header value. If you enter 0, the Expires header is not
included in the request.
The default is 30. Range: 0–(2
31
–1).
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.
The default is 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.
The default is 7200.
Reg Retry Intvl
Interval to wait before the Linksys IP phone retries registration after failing during the last
registration.
The default is 30.
Reg Retry Long Intvl
When registration fails with a SIP response code that does not match<Retry Reg RSC>,
the Linksys IP phone waits for the specified length of time before retrying. If this interval
is 0, the Linksys IP phone stops trying. This value should be much larger than the Reg
Retry Intvl value, which should not be 0.
The default is 1200.
Reg Retry Random Delay
Random delay range (in seconds) to add to <Register Retry Intvl> when retrying
REGISTER after a failure. This feature was added in Release 5.1.
The default is 0, which disables this feature.
Reg Retry Long Random Delay
Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying
REGSITER after a failure. This feature was added in Release 5.1.
The default is 0, which disables this feature.
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. This feature was added in Release 5.1.
The default value is 0, which disables the exponential back-off feature.
Page 98 / 186
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Document Version 3.2
Chapter 5
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SIP Tab
Response Status Code Handling
RTP Parameters
Field
Description
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 Linksys IP phone retries registration after failing during the last
registration.
The default is 30.
Field
Description
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.
The default is 16384.
RTP Port Max
Maximum port number for RTP transmission and reception.
The default is 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.
The default is 0.030.
Max RTP ICMP Err
Number of successive ICMP errors allowed when transmitting RTP packets to the peer
before the Linksys IP phone terminates the call. If value is set to 0, the Linksys IP phone
ignores the limit on ICMP errors.
The default is 0.
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SIP Tab
SDP Payload Types
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 Linksys IP phone 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/Linksys IP
phone-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the Linksys IP
phone’s local time, not the time reported by an NTP server. If the Linksys IP phone
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 Linksys IP phone web page.
The default is 0.
No UDP Checksum
Select yes if you want the Linksys IP phone to calculate the UDP header checksum for
SIP messages. Otherwise, select no.
The default is no.
Stats In BYE
Determines whether the Linksys IP phone 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>.
The default is no.
Field
Description
NSE Dynamic Payload
NSE dynamic payload type. The valid range is 96-127.
The default is 100.
AVT Dynamic Payload
AVT dynamic payload type. The valid range is 96-127.
The default is 101.
INFOREQ Dynamic Payload
INFOREQ dynamic payload type.
There is no default.
G726r16 Dynamic Payload
G.726-16 dynamic payload type. The valid range is 96-127.
The default is 98.
G726r24 Dynamic Payload
G.726-24 dynamic payload type. The valid range is 96-127.
The default is 97.
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Chapter 5
Linksys ATA Field Reference
SIP Tab
NAT Support Parameters
G726r40 Dynamic Payload
G.726-40 dynamic payload type. The valid range is 96-127.
The default is 96.
G729b Dynamic Payload
G.729b dynamic payload type. The valid range is 96-127.
The default is 99.
NSE Codec Name
NSE codec name used in SDP.
The default is NSE.
AVT Codec Name
AVT codec name used in SDP.
The default is telephone-event.
G711u Codec Name
G.711u codec name used in SDP.
The default is PCMU.
G711a Codec Name
G.711a codec name used in SDP.
The default is PCMA.
G726r16 Codec Name
G.726-16 codec name used in SDP.
The default is G726-16.
G726r24 Codec Name
G.726-24 codec name used in SDP.
The default is G726-24.
G726r32 Codec Name
G.726-32 codec name used in SDP.
The default is G726-32.
G726r40 Codec Name
G.726-40 codec name used in SDP.
The default is G726-40.
G729a Codec Name
G.729a codec name used in SDP.
The default is G729a.
G729b Codec Name
G.729b codec name used in SDP.
The default is G729ab.
G723 Codec Name
G.723 codec name used in SDP.
The default is G723.
EncapRTP Codec Name
EncapRTP codec name used in SDP.
The default is EncapRTP.
Field
Description

Rate

4 / 5 based on 1 vote.

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