SIP Profile Information
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Name
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Enter a name to identify the SIP profile; for example, SIP_7905. The value can include 1 to 50 characters, including alphanumeric
characters, dot, dash, and underscores.
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Description
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Identifies the purpose of the SIP profile. For example, SIP for 7970. The description can include up to 50 characters in any
language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>).
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Default MTP Telephony Event Payload Type
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Specifies the default payload type for RFC2833 telephony event. See RFC 2833 for more information. Usually, the default value
specifies the appropriate payload type. Ensure that you have a firm understanding of this parameter before changing it, as
changes could result in DTMF tones not being received or generated. The default value specifies 101 with range from 96 to
127.
The value of this parameter affects calls with the following conditions:
-
The call is an outgoing SIP call from Unified Communications Manager.
-
For the calling SIP trunk, the Media Termination Point Required check box is checked on the SIP Trunk Configuration window.
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Early Offer for G.Clear Calls
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The Early Offer for G.Clear Calls feature supports both standards-based G.Clear (CLEARMODE) and proprietary Cisco Session
Description Protocols (SDP).
To enable or disable Early Offer for G.Clear Calls, choose one of the following options:
-
Disabled
-
CLEARMODE
-
CCD
-
G.nX64
-
X-CCD
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SDP Session-level Bandwidth Modifier for Early Offer and Re-invites
|
Specifies the maximum amount of bandwidth that is needed when all the media streams are used. There are three Session Level
Bandwidth Modifiers: Transport Independent Application Specific (TIAS), Application Specific (AS), and Conference Total (CT).
Select one of the following options to specify which Session Level Bandwidth Modifier to include in the SDP portion of SIP
Early Offer or Reinvite requests.
-
TIAS and AS
-
TIAS only
-
AS only
-
CT only
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User-Agent and Server header information
|
Indicates how Unified Communications Manager handles the User-Agent and Server header information in a SIP message.
Choose one of the following three options:
-
Send Unified Communications Manager Version Information as User-Agent Header—For INVITE requests, the User-Agent header is
included with the CM version header information. For responses, the Server header is omitted. Unified Communications Manager passes through any contact headers untouched. This is the default behavior.
-
Pass Through Received Information as Contact Header Parameters—If this option is selected, the User-Agent/Server header information
is passed as Contact header parameters. The User-Agent/Server header is derived from the received Contact header parameters,
if present. Otherwise, they are taken from the received User-Agent/Server headers.
-
Pass Through Received Information as User-Agent and Server Header—If this option is selected, the User-Agent/Server header
information is passed as User-Agent/Server headers. The User-Agent/Server header is derived from the received Contact header
parameters, if present. Otherwise, they are taken from the received User-Agent/Server headers.
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Dial String Interpretation
|
Determine if the SIP identity header is a directory number or directory URI.
As directory numbers and directory URIs are saved in different database lookup tables, Unified Communications Manager examines
the characters in the SIP identity header's user portion, which is the portion of the SIP address that is before the @ sign
(for example, user@IP address or user@domain).
To configure the Dial String Interpretation, choose one of the following options from the list:
-
Always treat all dial strings as URI addresses—Unified Communications Manager treats the address of incoming calls as if they
were URI addresses.
-
Phone number consists of characters 0–9, A–D, *, and + (others that are treated as URI addresses)—Unified Communications Manager
treats the incoming call as a directory number if all the characters in the user portion of the SIP identity header fall within
this range. If the user portion of the address uses any characters that do not fall within this range, the address is treated
as a URI.
-
Phone number consists of characters 0–9, *, and + (others that are treated as URI addresses)—Unified Communications Manager
treats the incoming call as a directory number if all the characters in the user portion of the SIP identity header fall within
this range. If the user portion of the address uses any characters that do not fall within this range, the address is treated
as a URI.
Note
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If the user=phone tag is present in the Request URI, Unified Communications Manager always treats the dial string as a number
regardless of what option you choose for the Dial String Interpretation field.
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Accept Audio Codec Preferences in Received Offer
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Allows to select On to enable Unified Communications Manager to honor the preference of audio codecs in the received offer and preserve it while
processing. Select Off to enable Unified Communications Manager to ignore the preference of audio codecs in the received offer and apply the locally configured Audio Codec Preference List.
The default will select the service parameter configuration.
Note
|
If this is enabled in both incoming and outgoing trunks then the same codec preference list should be associated with both
trunks else it might result in a different codec being negotiated towards both sides leading to audio issues.
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Require SDP Inactive Exchange for Mid-Call Media Change
|
Designates how Unified Communications Manager handles mid-call updates to codecs or connection information such as IP address or port numbers.
If the check box is selcted, during mid-call codec or connection updates Unified Communications Manager sends an INVITE a=inactive SDP message to the endpoint to break the media exchange. This is required if an endpoint is not
capable of reacting to changes in the codec or connection information without disconnecting the media. This applies only to
audio and video streams within SIP-SIP calls.
If the check box is unchecked, Unified Communications Manager passes the mid-call SDP to the peer leg without sending a prior Inactive SDP to break the media exchange. This is the default
behavior.
Note
|
For early offer or best effort early offer enabled SIP trunks, this parameter will be overridden by the Send send-receive
SDP in mid-call INVITE parameter.
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|
Confidential Access Level Headers
|
Determines the inclusion of Confidential Access Level headers in INVITE and 200 OK messages. Valid values are as follows:
-
Disabled—CAL headers are not included.
-
Preferred—CAL headers are included and confidential-access-level tag is added in the Supported header.
-
Required— CAL headers are included and confidential-access-level tag is added in the Require and Proxy-Require headers.
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SDP Transparency Profile
|
Allows you to choose one of the following options for SIP profile :
-
None—Choose this option for Unified Communications Manager to filter out known SDP attributes only. By default, this option is
selected.
-
Pass all unknown SDP attributes—Choose this option for media adaptation and resilience (MARI). To ensure that the session level MARI attributes pass the
unknown attributes through Unified Communications Manager, choose this value on the SIP profile, which is associated with
both the originating device and the terminating device.
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Redirect by Application
|
Checking this check box and configuring this SIP Profile on the SIP trunk allows the Unified Communications Manager administrator to:
-
Apply a specific calling search space to redirected contacts that are received in the 3xx response.
-
Apply digit analysis to the redirected contacts to make sure that the call get routed correctly.
-
Prevent DOS attack by limiting the number of redirection (recursive redirection) that a service parameter can set.
-
Allow other features to be invoked while the redirection is taking place.
Getting redirected to a restricted phone number (such as an international number) means that handling redirection at the stack
level causes the call to be routed instead of being blocked. This behavior occurs if the Redirect by Application check box
is unchecked.
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Disable Early Media on 180
|
By default, Unified Communications Manager signals the calling phone to play local ringback if SDP is not received in the 180 response. If SDP is included in the 180
response, instead of playing ringback locally, Unified Communications Manager connects media, and the calling phone plays whatever the called device is sending (such as ringback or busy signal). If you
do not receive ringback, the device to which you are connecting may be including SDP in the 180 response, but it is not sending
any media before the 200OK response. In this case, check this check box to play local ringback on the calling phone and connect
the media upon receipt of the 200OK response
Note
|
Even though the phone that is receiving ringback is the calling phone, you need the configuration on the called device profile
because it determines the behavior.
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|
Outgoing T.38 INVITE Include Audio mline
|
Allows the system to accept a signal from Microsoft Exchange that causes it to switch the call from audio to T.38 fax. To
use this feature, you must also configure a SIP trunk with this SIP profile. For more information, see Chapter 68, Trunk "Configuration."
Note
|
The parameter applies to SIP trunks only, not phones that are running SIP or other endpoints.
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|
Offer valid IP and Send/Receive mode only for T.38 Fax Relay
|
If this checkbox is checked, this SIP profile on the trunk allows you to send a fax offer with a valid IP address and with
Send Receive SDP mode.
If this checkbox is not checked, this SIP profile on the trunk allows you to send a fax offer with a null IP address and with
Send Receive SDP mode.
This parameter applies only to trunks, not phones that are running SIP or other endpoints. It applies only for T38 fax relay
and, by default, this checkbox is unchecked.
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Enable ANAT
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Allows a dual-stack SIP trunk to offer both IPv4 and IPv6 media.
When you check both the Enable ANAT and the MTP Required check boxes, Unified Communications Manager inserts a dual-stack MTP and sends out an offer with two m-lines, one for IPv4 and another for IPv6. If a dual-stack MTP cannot be allocated, Unified Communications Manager sends an INVITE without SDP.
When you check the Enable ANAT check box and the Media Termination Point Required check box is unchecked, Unified Communications Manager sends an INVITE without SDP.
When the Enable ANAT and Media Termination Point Required check boxes display as unchecked (or when an MTP cannot be allocated),
Unified Communications Manager sends an INVITE without SDP.
When you uncheck the Enable ANAT check box but you check the Media Termination Point Required check box, consider the information,
which assumes that an MTP can be allocated:
-
Unified Communications Manager sends an IPv4 address in the SDP for SIP trunks with an IP Addressing Mode of IPv4 Only.
-
Unified Communications Manager sends an IPv6 address in the SDP for SIP trunks with an IP Addressing Mode of IPv6 Only.
-
For dual-stack SIP trunks, Unified Communications Manager determines which IP address type to send in the SDP based on the configuration for the IP Addressing Mode Preference for
Media enterprise parameter.
-
For dual-stack SIP trunks, Unified Communications Manager determines which IP address type to send in the SDP based on the configuration for the IP Addressing Mode Preference for
Media enterprise parameter.
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Require SDP Inactive Exchange for Mid-Call Media Change
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Designates how Unified Communications Manager handles mid-call updates to codecs or connection information such as IP address or port numbers.
If the box is checked, during mid-call codec or connection updates Unified Communications Manager sends an INVITE a=inactive SDP message to the endpoint to break the media exchange. This is required if an endpoint is not
capable of reacting to changes in the codec or connection information without disconnecting the media. This applies only to
audio and video streams within SIP-SIP calls.
Note
|
For early offer enabled SIP trunks, this parameter will be overridden by the Send send-receive SDP in mid-call INVITE parameter.
|
If the box is unchecked, Unified Communications Manager passes the mid-call SDP to the peer leg without sending a prior Inactive SDP to break the media exchange. This is the default
behavior.
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Use Fully Qualified Domain Name in SIP Requests
|
Enables Unified Communications Manager to relay an alphanumeric hostname of a caller by passing it through to the called device or outbound trunk as a part of the
SIP header information.
-
If the box is unchecked, the IP address for Unified Communications Manager will be passed to the line device or outbound trunk instead of the user’s hostname. This is the default behavior.
-
If the box is checked, Unified Communications Manager will relay an alphanumeric hostname of a caller by passing it through to the called endpoint as a part of the SIP header
information. This enables the called endpoint to return the call using the received or missed call list. If the call is originating
from a line device on the Unified Communications Manager cluster, and is being routed on a SIP trunk then the configured Organizational Top-Level Domain (e.g., cisco.com) will be
used in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID. If the call is originating from a trunk on
Unified Communications Manager and is being routed on a SIP trunk then:
-
If the inbound call provides a host or domain in the caller’s information, the outbound SIP trunk messaging will preserve
the hostname in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID
-
If the inbound call does not provide a host or domain in the caller's information, the configured Organizational Top-Level
Domain will be used in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID
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Assured Services SIP conformance
|
Specifies to check this box for third-party AS-SIP endpoints as well as AS-SIP trunks to ensure proper Assured Service behavior.
This setting provides specific Assured Service behavior that affects services such as Conference factory and SRTP.
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Enable External QoS
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Specifies to check this box to configure this SIP Profile for external QoS support. With this feature enabled, you can use
an APIC-EM Controller to manage QoS for SIP media flows for devices that use this SIP Profile. The default value is unchecked.
Note
|
This check box appears only if the External QoS Enable service parameter is set to True.
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Parameters Used in Phone
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Timer Invite Expires (seconds)
|
Sspecifies the time, in seconds, after which a SIP INVITE expires. The Expires header uses this value. Valid values include
any positive number; 180 specifies the default.
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Timer Register Delta (seconds)
|
Intended to be used by SIP endpoints only. The endpoint receives this value via a tftp config file. The end point reregisters
Timer Register Delta seconds before the registration period ends. The registration period gets determined by the value of
the SIP Station KeepAlive Interval service parameter. Valid values for Timer Register Delta range from 32767 to 0. The default
value is 5.
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Timer Register Expires (seconds)
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Intended to be used by SIP endpoints only. The SIP endpoint receives the value via a tftp config file. This field specifies
the value that the phone that is running SIP sends in the Expires header of the REGISTER message. Valid values include any
positive number; however, 3600 (1 hour) specifies the default value.
If the endpoint sends a shorter Expires value than the value of the SIP Station Keepalive Interval service parameter, Unified
Communications Manager responds with a 423 "Interval Too Brief".
If the endpoint sends an Expires value that is greater than the SIP Station Keepalive Interval service parameter value, Unified
Communications Manager responds with a 200 OK that includes the Keepalive Interval value for Expires.
Note
|
For mobile phones that are running SIP, Unified Communications Manager uses the value in this field instead of the value that the SIP Station KeepAlive Interval service parameter specifies to
determine the registration period.
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Note
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For TCP connections, the value for the Timer Register Expires field must be lower than the value for the SIP TCP Unused Connection
service parameter.
|
|
Timer T1 (msec)
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Specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive
number. Default specifies 500.
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Timer T2 (msec)
|
Specifies the highest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive
number. Default specifies 4000.
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Retry INVITE
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Specifies the maximum number of times that an INVITE request gets retransmitted. Valid values include any positive number.
Default specifies 6.
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Retry Non-INVITE
|
Specifies the maximum number of times that a SIP message other than an INVITE request gets retransmitted. Valid values include
any positive number. Default specifies 10.
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Media Port Ranges
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Specifies to click the radio button that corresponds to how you want to manage QoS for audio and video calls for devices that
are associated to this SIP Profile
-
Common Port Range for Audio and Video—Choose this option if you want to use a common port range that can handles both the audio and video media stream.
-
Separate Port Ranges for Audio and Video—Choose this option if you want to set up a distinct port range for the audio stream and a distinct port range for the video
stream.
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Start Media Port
|
Designates the start real-time protocol (RTP) port for media. Media port ranges from 2048 to 65535. Default specifies 16384.
This field appears when you select Common Port Range for Audio and Video as the Media Port Range.
|
Stop Media Port
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Designates the stop real-time protocol (RTP) port for media. Media port ranges from 2048 to 65535. Default specifies 32766.
This field appears when you select Common Port Range for Audio and Video for the Media Port Range.
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Start Audio Port
|
Allows you to create a port range for audio by entering the start of the port range. For example, 16384. The audio port range
cannot overlap the video port range.
This field appears when you select Separate Port Ranges for Audio and Video for the Media Port Range.
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Stop Audio Port
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Allows you to enter the ending of the port range for audio calls. The audio port range must not overlap the video port range.
For example, 32766.
This field appears when you select Separate Port Ranges for Audio and Video for the Media Port Range.
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Start Video Port
|
Allows you to create a port range for the video stream of a video call by entering the beginning of the port range. For example,
32767. The video port range cannot overlap with the audio port range.
This field appears when you select Separate Port Ranges for Audio and Video for the Media Port Range.
|
Stop Video Port
|
Allows you to enter the ending of the port range for audio calls. The audio port range must not overlap the video port range.
This field appears when you select Separate Port Ranges for Audio and Video for the Media Port Range.
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DSCP for Audio Calls
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Allows you to select the value that you want to assign as the DSCP value for audio-only calls. The Default Option is to use
the value of the DSCP for Audio Calls service parameter.
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DSCP for Video Calls
|
Allows you to select the value that you want to assign as the DSCP value for video calls. The Default Option is to use the
value of the DSCP for Video Calls service parameter.
|
DSCP for Audio Portion of Video Calls
|
Allows you to select the value that you want to assign as the DSCP value for audio portion of a video call. The default option
is to use the value that is configured in the DSCP for Audio Portion of Video Calls service parameter.
Note
|
If you choose a different DSCP value for audio portion of video calls than you configured for DSCP Video Calls, it could mean
that the audio and video streams within a single video call could have different DSCP markings and different QoS policy control,
which could result in lip sync issues that result from network bandwidth issues.
|
|
DSCP for TelePresence Calls
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Allows you to select the value that you want to assign as the DSCP value for TelePresence calls. The default option is to
use the value of the DSCP for TelePresence Calls service parameter.
|
DSCP for Audio Portion of TelePresence Calls
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Allows you to select the value that you want to assign as the DSCP value for the audio portion of TelePresence calls. The
default option is to use the value of the DSCP for TelePresence Calls service parameter.
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Call Pickup URI
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Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the call pickup feature.
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Call Pickup Group Other URI
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Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the call pickup group other feature.
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Call Pickup Group URI
|
Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the call pickup group feature.
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Meet Me Service URI
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Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the meet me conference feature.
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User Info
|
Configures the user= parameter in the REGISTER message.
Valid values follow:
-
none—No value gets inserted.
-
phone—The value user=phone gets inserted in the To, From, and Contact Headers for REGISTER.
-
ip—The value user=ip gets inserted in the To, From, and Contact Headers for REGISTER.
|
DTMF DB Level
|
Specifies in-band DTMF digit tone level. Valid values follow:
-
1 to 6 dB below nominal
-
2 to 3 dB below nominal
-
3 nominal
-
4 to 3 dB above nominal
-
5 to 6 dB above nominal
|
Call Hold Ring Back
|
Indicates the call on hold status. For example, if you have a call on hold and are talking on another call, when you hang
up the call, this parameter causes the phone to ring to let you know that you still have another party on hold. Valid values
follow:
|
Anonymous Call Block
|
Configures anonymous call block. Valid values follow:
|
Caller ID Blocking
|
Configures caller ID blocking. When blocking is enabled, the phone blocks its own number or e-mail address from phones that
have caller identification enabled. Valid values follow:
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Do Not Disturb Control
|
Sets the Do Not Disturb (DND) feature. Valid values follow:
|
Telnet Level for 7940 and 7960
|
Cisco Unified IP Phones 7940 and 7960 do not support ssh for login access or HTTP that is used to collect logs; however, these phones support Telnet,
which lets the user control the phone, collect debugs, and look at configuration settings. This field controls the telnet_level
configuration parameter with the following possible values:
|
Resource Priority Namespace
|
Enables the admin to select one of the cluster’s defined Resource Priority Namespace network domains for assignment to a line
via its SIP Profile.
|
Timer Keep Alive Expires (seconds)
|
Specifies the interval between keepalive messages that are sent to the backup Unified Communications Manager to ensure that it is available in the event that a failover is required.
Unified Communications Manager requires a keepalive mechanism to support redundancy.
|
Timer Subscribe Expires (seconds)
|
Specifies the time, in seconds, after which a subscription expires. This value gets inserted into the Expires header field.
Valid values include any positive number; however, 120 specifies the default value.
|
Timer Subscribe Delta (seconds)
|
Allows you to use this parameter in conjunction with the Timer Subscribe Expires setting. The phone resubscribes Timer Subscribe
Delta seconds before the subscription period ends, as governed by Timer Subscribe Expires. Valid values range from 3 to 15.
Default specifies 5.
|
Maximum Redirections
|
Allows you to use this configuration variable to determine the maximum number of times that the phone allows a call to be
redirected before dropping the call. Default specifies 70 redirections.
|
Off Hook to First Digit Timer (microseconds)
|
Specifies the time in microseconds that passes when the phone goes off hook and the first digit timer gets set. The value
ranges from 0 - 150,000 microseconds. Default specifies 15,000 microseconds.
|
Call Forward URI
|
Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the call forward feature.
|
Abbreviated Dial URI
|
Provides a unique address that the phone that is running SIP sends to Unified Communications Manager to invoke the abbreviated dial feature.
Speed dials that are not associated with a line key (abbreviated dial indices) do not download to the phone. The phone uses
the feature indication mechanism (INVITE with Call-Info header) to indicate when an abbreviated dial number has been entered.
The request URI contains the abbreviated dial digits (for example, 14), and the Call-Info header indicates the abbreviated
dial feature. translates the abbreviated dial digits into the configured digit string and extend the call with that string.
If no digit string has been configured for the abbreviated dial digits, a 404 Not Found response gets returned to the phone.
|
Conference Join Enabled
|
Determines whether the Unified Communications Managers 7940 or 7960, when the conference initiator that is using that phone hangs up, should attempt to join the remaining conference
attendees. Check the check box if you want to join the remaining conference attendees; leave it unchecked if you do not want
to join the remaining conference attendees.
Note
|
This check box applies to the IM and Presence Services 7941/61/70/71/11 when they are in SRST mode only.
|
|
RFC 2543 Hold
|
Enables setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Unified Communications Manager. This allows backward compatibility with endpoints that do not support RFC3264.
|
Semi Attended Transfer
|
Determines whether the Cisco Unified IP Phones 7940 and 7960 caller can transfer the second leg of an attended transfer while the call is ringing. Check the check box
if you want semi-attended transfer enabled; leave it unchecked if you want semi-attended transfer disabled.
Note
|
This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.
|
|
Enable VAD
|
Enables Voice Activation Detection (VAD). When VAD is enabled, media is not transmitted until the voice is detected.
|
Stutter Message Waiting
|
Enables stutter dial tone when the phone goes off hook and a message is waiting; leave unchecked if you do not want a stutter
dial tone when a message is waiting.
This setting supports Cisco Unified IP Phones 7960 and 7940 that run SIP.
|
MLPP User Authorization
|
Enable MLPP User Authorization. MLPP User Authorization requires the phone to send in an MLPP username and password.
|
Normalization Script
|
Normalization Script
|
Alows you to choose the script that you want to apply to this SIP profile.
To import another script, go to the SIP Normalization Script Configuration window (), and import a new script file.
Caution
|
A normalization script in the SIP profile is only valid for non-trunk devices.
|
|
Parameter Name/Parameter Value
|
Optionally, enter parameter names and parameter values. Valid values include all characters except equals signs (=), semi-colons
(;), and non-printable characters, such as tabs. You can enter a parameter name with no value.
To add another parameter line, click the + (plus) button. To delete a parameter line, click the - (minus) button.
Note
|
You must choose a script from the Normalization Script list before you can enter parameter names and values.
|
|
Enable Trace
|
Enables tracing within the script or uncheck this check box to disable tracing. When checked, the trace.output API provided
to the Lua scripter produces SDI trace
Note
|
We recommend that you only enable tracing while debugging a script. Tracing impacts performance and should not be enabled
under normal operating conditions.
|
|
Incoming Requests FROM URI Settings
|
Caller ID DN
|
Allows you to enter the pattern that you want to use for calling line ID, from 0 to 24 digits. For example, in North America:
-
555XXXX = Variable calling line ID, where X equals an extension number. The CO appends the number with the area code if you
do not specify it.
-
55000 = Fixed calling line ID, where you want the Corporate number to be sent instead of the exact extension from which the
call is placed. The CO appends the number with the area code if you do not specify it.
You can also enter the international escape character +.
|
Caller Name
|
Allows you to enter a caller name to override the caller name that is received from the originating SIP Device.
|
Trunk Specific Configuration
|
Reroute Incoming Request to new Trunk based on
|
Unified Communications Manager only accepts calls from the SIP device whose IP address matches the destination address of the configured SIP trunk. In addition,
the port on which the SIP message arrives must match the one configured on the SIP trunk. After the Unified Communications Manager accepts the call, it uses the configuration for this setting to determine whether the call should get rerouted to another
trunk.
You can select any of the following method that Unified Communications Manager uses to identify the SIP trunk where the call is rerouted:
-
Never—If the SIP trunk matches the IP address of the originating device, choose this option, which equals the default setting.
The Unified Communications Manager identifies the trunk by using the source IP address of the incoming packet and the signaling port number, do not route the
call to a different (new) SIP trunk. The call occurs on the SIP trunk on which the call arrived.
-
Contact Info Header—If the SIP trunk uses a SIP proxy, choose this option. The Unified Communications Manager analyzes the contact header in the incoming request and uses the IP address or domain name and signaling port number that
is specified in the header to reroute the call to the SIP trunk that uses the IP address and port number. If no SIP trunk
is identified, the call occurs on the trunk on which the call arrived.
-
Call-Info Header with purpose=x-cisco-origIP—If the SIP trunk uses a Customer Voice Portal (CVP) or a Back-to-Back User Agent
(B2BUA), choose this option. When the incoming request is received, the Unified Communications Manager analyzes the Call-Info header, search for the parameter purpose=x-cisco-origIP, and uses the IP address or domain name specified
in the header to reroute the call to the SIP trunk that uses the IP address and port. The listening port on the inbound trunk
and the trunk targeted by the x-cisco-origIP value need to match for the targeted trunk to be used in the call . If the parameter
does not exist in the header or no SIP trunk is identified, the call occurs on the SIP trunk on which the call arrived.
Note
|
You cannot set these parameters as they are not supported in Unified Communications Manager for secure calls.
This setting does not work for SIP trunks that are connected to a IM and Presence Service proxy server or SIP trunks that are connected to originating gateways in different Unified CM groups.
|
|
Resource Priority Namespace List
|
Allows you to select a configured Resource Priority Namespace list. Configure the lists in the Resource Priority Namespace
List menu that is accessed from .
|
SIP Rel1XX Options
|
Configures SIP Rel1XX, which determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably
to the remote SIP endpoint. Valid values follow:
-
Disabled—Disables SIP Rel1XX.
-
Send PRACK if 1XX contains SDP—Acknowledges a 1XX message with PRACK, only if the 1XX message contains SDP.
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Send PRACK for all 1XX messages—Acknowledges all1XX messages with PRACK.
Note
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You need not configure the above field if Connect Inbound Call before Playing Queuing Announcement checkbox is checked in the Trunk Specific Configuration.
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Session Refresh Method
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Session Timer with Update: The session refresh timer allows for periodic refresh of SIP sessions, which allows the Unified Communications Manager and remote agents to determine whether the SIP session is still active. Prior to Release 10.01, when the Unified Communications Manager received a refresh command, it supported receiving either Invite or Update SIP requests to refresh the session. When the
Unified Communications Manager initiated a refresh, it supported sending only Invite SIP requests to refresh the session. With Release 10.01, this feature
extends the refresh capability so that Unified Communications Manager can send both Update and Invite requests.
Specify whether Invite or Update should be used as the Session Refresh Method.
Invite (default):
Note
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Sending a mid-call Invite request requires that an offer SDP be specified in the request. This means that the far end must
send an answer SDP in the Invite response.
Update: Unified Communications Manager sends a SIP Update request, if support for the Update method is specified by the far end of the SIP session either in the
Supported or Require headers. When sending the Update request, the Unified Communications Manager includes an SDP. This simplifies the session refresh since no SDP offer/answer exchange is required.
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Note
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If the Update method is not supported by the far end of the SIP session, the Unified Communications Manager continues to use the Invite method for session refresh.
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Early Offer support for voice and video calls
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Configures Early Offer support for voice and video calls. When enabled, Early Offer support includes a session description
in the initial INVITE for outbound calls. Early Offer configuration settings on SIP profile apply only to SIP trunk calls.
These configuration settings do not affect SIP line side calls. If this profile is shared between a trunk and a line, only
a SIP trunk that uses the profile is affected by these settings.
The Media Transfer Point (MTP) Required check box on the Trunk Configuration window, if enabled, overrides the early offer
configuration on the associated SIP profile. Unified Communications Manager sends the MTP IP address and port with a single
codec in the SDP in the initial INVITE.
Select one of the following three options:
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Disabled (Default value) - Disables Early Offer; no SDP will be included in the initial INVITE for outbound calls.
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Best Effort (no MTP inserted)
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Provide Early Offer for the outbound call only when caller side's media port, IP and codec information is available.
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Provide Delayed Offer for the outbound call when caller side's media port, IP and codec information is not available. No MTP
is inserted to provide Early Offer in this case.
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Mandatory(insert MTP if needed) - Provide Early Offer for all outbound calls and insert MTP when caller side's media port,
IP and codec information is not available.
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Video Call Traffic Class
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Determines the type of video endpoint or trunk that the SIP Profile is associated with. From the list, select one of the
following three options
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Immersive—High-definition immersive video.
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Desktop—Standard desktop video.
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Mixed—A mix of immersive and desktop video.
Unified Communications Manager Locations Call Admission Control (CAC) reserves bandwidth from two Locations video bandwidth
pools, "Video Bandwidth" and/or "Immersive Bandwidth", depending on the type of call determined by the Video Call Traffic
Class.
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Calling Line Identification Presentation
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Select Strict From URI presentation Only to select the network provided identity.
Select Strict Identity Headers presentation Only to select the user provided identity.
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Deliver Conference Bridge Identifier
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Allows the SIP trunk to pass the b-number that identifies the conference bridge across the trunk instead of changing the b-number
to the null value.
The terminating side does not require that this field be enabled.
Checking this check box is not required for Open Recording Architecture (ORA) SIP header enhancements to the Recording feature
to work.
Enabling this check box allows the recorder to coordinate recording sessions where the parties are participating in a conference.
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Early Offer support for voice and video calls (insert MTP if needed)
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Allows you want to create a trunk that supports early offer.
Early Offer configurations on SIP profile apply to SIP trunk calls. These configurations do not affect SIP line side calls.
If this profile is shared between a trunk and a line, only the SIP trunk that uses the profile provides early offer.
Note
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When checked, the Media Termination Required check box on the Trunk Configuration window overrides the early offer configuration
on the associated SIP profile. The Unified Communications Manager sends the MTP IP address and port with a single codec in the SDP in the initial INVITE.
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Send send-receive SDP in mid-call INVITE
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Allows you to prevent Unified Communications Manager from sending an INVITE a=inactive SDP message during call hold or media break during supplementary services.
Note
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This check box applies only to early offer or best early offer enabled SIP trunks and has no impact on SIP line calls.
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When you enable Send send-receive SDP in mid-call INVITE for an early offer or best early offer SIP trunk in tandem mode,
Unified Communications Manager inserts MTP to provide sendrecv SDP when a SIP device sends offer SDP with a=inactive or sendonly or recvonly in audio media
line. In tandem mode, depends on the SIP devices to initiate reestablishment of media path by sending either a delayed INVITE
or mid-call INVITE with send-recv SDP.
When you enable both Send send-receive SDP in mid-call INVITE and Require SDP Inactive Exchange for Mid-Call Media Change
on the same SIP Profile, the Send send-receive SDP in mid-call INVITE overrides the Require SDP Inactive Exchange for Mid-Call
Media Change, so Unified Communications Manager does not send an INVITE with a=inactive SDP in mid-call codec updates. For SIP line side calls, the Require SDP Inactive
Exchange for Mid-Call Media Change check box applies when enabled.
Note
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To prevent the SDP mode from being set to inactive in a multiple-hold scenario, set the Duplex Streaming Enabled clusterwide
service parameter () to True.
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Allow Presentation Sharing using BFCP
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Allows the supported SIP endpoints to use the Binary Floor Control Protocol to enable presentation sharing.
The use of BFCP creates an additional media stream in addition to the existing audio and video streams. This additional stream
is used to stream a presentation, such as a PowerPoint presentation from someone’s laptop, into a SIP videophone.
If the box is unchecked, Unified Communications Manager rejects BFCP offers from devices associated with the SIP profile by setting the BFCP application line and associated media
line ports to 0 in the answering SDP message. This is the default behavior.
Note
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BFCP is only supported on SIP networks. BFCP must be enabled on all SIP trunks, lines, and endpoints for presentation sharing
to work. BFCP is not supported if the SIP line or SIP trunk uses MTP, RSVP, TRP or Transcoder.
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Allow iX Application Media
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Enables support for iX media channel.
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Allow Passthrough of Configured Line Device Caller Information
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Allows passthrough of configured line device caller information from the SIP trunk.
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Reject Anonymous Incoming Calls
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Allows to reject anonymous incoming calls.
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Reject Anonymous Outgoing Calls
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Allows to reject anonymous outgoing calls.
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Allow multiple codecs in answer SDP
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Applies when incoming SIP signals do not indicate support for multiple codec negotiation and Unified Communications Manager can finalize the negotiated codec.
When this check box is checked, the endpoint behind the trunk is capable of handling multiple codecs in the answer SDP.
For example, an endpoint that supports multiple codec negotiation calls the SIP trunk and Unified Communications Manager sends a Delay Offer request to a trunk. The endpoint behind the trunk returns all support codecs without the Contact header
to indicate the support of multiple codec negotiation.
In this case, Unified Communications Manager identifies the trunk as capable of multiple codec negotiation and sends SIP response messages back to both endpoints with
multiple common codecs.
When this check box is unchecked, Unified Communications Manager identifies the endpoint behind the trunk as incapable of multiple codec negotiation, unless indicated otherwise by SIP contact
header URI. Unified Communications Manager continues the call with single codec negotiation.
Configure Allow multiple codecs in answer SDP for the following:
Do not configure this capability for SIP intercluster trunks to Cisco SME or other Unified Communications Manager systems.
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Send ILS Learned Destination Route String
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Allows the calls that Unified Communications Manager routes to a learned directory URI, learned number, or learned pattern,
Unified Communications Manager adds the x-cisco-dest-route-string header to outgoing SIP INVITE and SUBSCRIBE messages and inserts the destination route string into the header.
When this check box is unchecked, Unified Communications Manager does not add the x-cisco-dest-route-string header to any SIP messages.
The x-cisco-dest-route-string header allows Unified Communications Manager to route calls across a Unified Border Element.
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Connect Inbound Call before Playing Queuing Announcement
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Allows you to send the carrier a CONNECT message before playing the hunt group announcements. You should enable this feature
if the carrier trunk does not support in-band call status updates or if external callers report that they are unable to hear
hunt group announcements.
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SIP OPTIONS Ping
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Enable OPTIONS Ping to monitor destination status for Trunks with service type "None (Default)"
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Allows you to enable the SIP OPTIONS feature.
SIP OPTIONS are requests to the configured destination address on the SIP trunk. If the remote SIP device fails to respond
or sends back a SIP error response such as 503 Service Unavailable or 408 Timeout, Unified Communications Manager reroute the calls using other trunks or using a different address.
The OPTIONS ping interval value for In-service and Partially In-service ranges from 5 to 600 seconds. The default value is
60 seconds. A SIP trunk is set to In-service when it receives a success response from the peer. If the peer fails to respond
due to some errors, then the status is set to Out-of-service. The SIP trunk does not know the peer status until the next time
OPTIONS ping is sent.
If the SIP trunk sends any message between the ping interval and if the peer destination is Out-of service because of any
error, the message results in failure. You can change the ping timer to a smaller value if required.
If this check box is unchecked, the SIP trunk does not track the status of SIP trunk destinations.
If this check box is checked, you can change the ping timer to a smaller value if required.
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Ping Interval for In-service and Partially In-service Trunks (seconds)
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Configures the time duration between SIP OPTIONS requests when the remote peer is responding and the trunk is marked as In
Service. If at least one IP address is available, the trunk is In Service; if all IP addresses are unavailable, the trunk
is Out of Service.
The default value specifies 60 seconds. Valid values range from 5 to 600 seconds.
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Ping Interval for Out-of-service SIP Trunks (seconds)
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Configures the time duration between SIP OPTIONS requests when the remote peer is not responding and the trunk is marked as
Out of Service. The remote peer may be marked as Out of Service if it fails to respond to OPTIONS, if it sends 503 or 408
responses, or if the Transport Control Protocol (TCP) connection cannot be established. If at least one IP address is available,
the trunk is In Service; if all IP addresses are unavailable, the trunk is Out of Service.
The default value specifies 120 seconds. Valid values range from 5 to 600 seconds.
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Ping Retry Timer (milliseconds)
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Specifies the maximum waiting time before retransmitting the OPTIONS request.
Valid values range from 100 to 1000 milliseconds. The default value specifies 500 milliseconds.
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Ping Retry Count
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Specifies the number of times that Unified Communications Manager resends the OPTIONS request to the remote peer. After the configured retry attempts are used, the destination is considered
to have failed. To obtain faster failure detection, keep the retry count low.
Valid values range from 1 to 10. The default value specifies 6.
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