Cisco Unified Communications Manager Administration Guide, Release 8.5(1)
SIP Profile Configuration
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SIP Profile Configuration

Table Of Contents

SIP Profile Configuration

SIP Profile Configuration Settings

Synchronizing a SIP Profile With Affected SIP Devices

Related Topics


SIP Profile Configuration


A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.

Use the following topics to configure and locate SIP profiles:

SIP Profile Configuration Settings

Synchronizing a SIP Profile With Affected SIP Devices

Related Topics

SIP Profile Configuration Settings

A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.

Tips About Resetting a SIP Profile

For instructions on how to reset a SIP profile, see the descriptions of the Reset Selected and Reset buttons in the "Common Buttons and Icons" section on page 1-18 of the Cisco Unified Communications Manager System Guide.

You may also want to consult the following topic:

Synchronizing a SIP Profile With Affected SIP Devices

Tips About Deleting SIP Profiles

To find out which devices are using the SIP profile, choose Dependency Records link from the Related Links drop-down list box in the SIP Profile Configuration window. If the dependency records are not enabled for the system, the dependency records summary window displays a message. For more information about dependency records, see the "Accessing Dependency Records" section on page A-2 of the Cisco Unified Communications Manager System Guide.

Using the GUI

For instructions on how to use the Cisco Unified Communications Manager Administration Graphical User Interface (GUI) to find, delete, configure, or copy records, see the "Navigating the Cisco Unified Communications Manager Administration Application" section on page 1-13 and its subsections, which explain how to use the GUI and detail the functions of the buttons and icons.

Configuration Settings Table

Table 76-1 describes the available settings in the SIP Profile Configuration window. For more information about related procedures, see the "Related Topics" section.

Table 76-1 SIP Profile Configuration Settings 

Field
Description
SIP Profile Information

Name

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.

Description

This field 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 (<>).

Default MTP Telephony Event Payload Type

This field specifies the default payload type for RFC2833 telephony event. See RFC 2833 for more information. In most cases, the default value specifies the appropriate payload type. Be sure 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 Cisco Unified Communications Manager.

For the calling SIP trunk, the Media Termination Point Required check box is checked on the SIP Trunk Configuration window.

Resource Priority Namespace List

Select a configured Resource Priority Namespace Network Domain list from the drop-down menu. Configure the lists in the Resource Priority Namespace Network Domain menu that is accessed from the System menu.

Early Offer for G.Clear Calls

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

Redirect by Application

Checking this check box and configuring this SIP Profile on the SIP trunk allows the Cisco 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.

See the "Redirection" section on page 40-20.

Disable Early Media on 180

By default, Cisco Unified Communications Manager signals the calling phone to play local ringback if SDP is not received in the 180 or 183 response. If SDP is included in the 180 or 183 response, instead of playing ringback locally, Cisco 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.

See the "Use of Early Media" section on page 40-14.

Outgoing T.38 INVITE Include Audio mline

The parameter 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.

Enable ANAT

This option 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, Cisco 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, Cisco 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, Cisco 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), Cisco 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:

Cisco Unified Communications Manager sends an IPv4 address in the SDP for SIP trunks with an IP Addressing Mode of IPv4 Only.

Cisco 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, Cisco 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.

 

Parameters used in Phone

Timer Invite Expires (seconds)

This field specifies 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.

Timer Register Delta (seconds)

Use this parameter in conjunction with the Timer Register Expires setting. The phone 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 range from 32767 to 0. Default specifies 5.

Timer Register Expires (seconds)

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. In the 200OK response to REGISTER, Cisco Unified Communications Manager includes an Expires header with the configured value of the SIP Station KeepAlive Interval service parameter. This value in the 200OK determines the time, in seconds, after which the registration expires. The phone refreshes the registration Timer Register Delta seconds before the end of this interval.

Note For dual-mode phones that are running SIP, Cisco 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.

Timer T1 (msec)

This field specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 500.

Timer T2 (msec)

This field specifies the highest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 4000.

Retry INVITE

This field specifies the maximum number of times that an INVITE request gets retransmitted. Valid values include any positive number. Default specifies 6.

Retry Non-INVITE

This field 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.

Start Media Port

This field designates the start real-time protocol (RTP) port for media. Media port ranges from 16384 to 32766. Default specifies 16384.

Stop Media Port

This field designates the stop real-time protocol (RTP) port for media. Media port ranges from 16384 to 32766. Default specifies 32766.

Call Pickup URI

This URI provides a unique address that the phone that is running SIP sends to Cisco Unified Communications Manager to invoke the call pickup feature.

Call Pickup Group Other URI

This URI provides a unique address that the phone that is running SIP sends to Cisco Unified Communications Manager to invoke the call pickup group other feature.

Call Pickup Group URI

This URI provides a unique address that the phone that is running SIP sends to Cisco Unified Communications Manager to invoke the call pickup group feature.

Meet Me Service URI

This URI provides a unique address that the phone that is running SIP sends to Cisco Unified Communications Manager to invoke the meet me conference feature.

User Info

This field 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

This field 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

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:

Off permanently and cannot be turned on and off locally by using the user interface.

On permanently and cannot be turned on and off locally by using the user interface.

Anonymous Call Block

This field configures anonymous call block. Valid values follow:

Off—Disabled permanently and cannot be turned on and off locally by using the user interface.

On—Enabled permanently and cannot be turned on and off locally by using the user interface.

Caller ID Blocking

This field 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:

Off—Disabled permanently and cannot be turned on and off locally by using the user interface.

On—Enabled permanently and cannot be turned on and off locally by using the user interface.

Do Not Disturb Control

This field sets the Do Not Disturb (DND) feature. Valid values follow:

User—The dndControl parameter for the phone should specify 0.

Admin—The dndControl parameter for the phone should specify 2.

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:

Disabled (no access)

Limited (some access but cannot run privileged commands)

Enabled (full access)

Timer Keep Alive Expires (seconds)

Cisco Unified Communications Manager requires a keepalive mechanism to support redundancy. This field specifies the interval between keepalive messages that are sent to the backup Cisco Unified Communications Manager to ensure that it is available in the event that a failover is required.

Timer Subscribe Expires (seconds)

This field 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)

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

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)

This field 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 - 15,000 microseconds. Default specifies 15,000 microseconds.

Call Forward URI

This URI provides a unique address that the phone that is running SIP sends to Cisco Unified Communications Manager to invoke the call forward feature.

Abbreviated Dial URI

This URI provides a unique address that the phone that is running SIP sends to Cisco 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. Cisco Unified Communications Manager 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

This check box determines whether the Cisco Unified IP Phones 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 Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.

RFC 2543 Hold

Check this check box to enable setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Cisco Unified Communications Manager. This allows backward compatibility with endpoints that do not support RFC3264.

Semi Attended Transfer

This check box 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

Check this check box if you want voice activation detection (VAD) enabled; leave it unchecked if you want VAD disabled. When VAD is enabled, no media gets transmitted when voice is detected.

Stutter Message Waiting

Check this check box if you want 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.

Call Stats

Check this check box if you want RTP statistics in BYE requests and responses enabled; leave unchecked if you want RTP statistics in BYE requests and responses disabled.

If this check box is checked, the phone inserts the headers RTP-RxStat and RTP-TxStat as follows:

RTP-RxStat:Dur=a,Pkt=b,Oct=c,LatePkt=d,LostPkt=e,AvgJit=f

RTP-TxStat: Dur=g,Pkt=h,Oct=i

where:

Dur—Total number of seconds since the beginning of reception or transmission.

Pkt—Total number of RTP packets that are received or transmitted.

Oct—Total number of RTP payload octets that are received or transmitted (not including RTP header).

LatePkt—Total number of late RTP packets that are received.

LostPkt—Total number of lost RTP packets that are received (not including the late RTP packets).

AvgJit—Average jitter, which is an estimate of the statistical variance of the RTP packet interarrival time, measured in timestamp unit and calculated according to RFC 1889.

a, b, c, d, e, f, g, h, and i—Integers

Trunk Specific Configuration

Reroute Incoming Request to new Trunk based on

Cisco 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 that is configured on the SIP trunk. After Cisco Unified Communications Manager accepts the call, Cisco Unified Communications Manager uses the configuration for this setting to determine whether the call should get rerouted to another trunk.

From the drop-down list box, choose the method that Cisco Unified Communications Manager uses to identify the SIP trunk where the call gets rerouted:

Never—If the SIP trunk matches the IP address of the originating device, choose this option, which equals the default setting. Cisco Unified Communications Manager, which identifies the trunk by using the source IP address of the incoming packet and the signaling port number, does 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. Cisco Unified Communications Manager parses 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. 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, Cisco Unified Communications Manager parses the Call-Info header, looks for the parameter, purpose=x-cisco-origIP, and uses the IP address or domain name and the signaling port number that is specified in the header to reroute the call to the SIP trunk that uses the IP address and port. 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.

Tip This setting does not work for SIP trunks that are connected to a Cisco Unified Presence proxy server or SIP trunks that are connected to originating gateways in different Cisco Unified CM groups.

RSVP Over SIP

This field configures RSVP over SIP trunks. From the drop-down list box, choose the method that Cisco Unified Communications Manager uses to configure RSVP over SIP trunks:

Local RSVP—In a local configuration, RSVP occurs within each cluster, between the end point and the local SIP trunk, but not on the WAN link between the clusters.

E2E—In an end-to-end (E2E) configuration, RSVP occurs on the entire path between the end points, including within the local cluster and over the WAN.

Fall back to local RSVP

Check this box if you want to allow failed end-to-end RSVP calls to fall back to local RSVP to establish the call. If this box is not checked, end-to-end RSVP calls that cannot establish an end-to-end connection fail.

SIP Rel1XX Options

This field 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.

Send PRACK for all 1XX messages—Acknowledges all1XX messages with PRACK.

If you set the RSVP Over SIP field to E2E, you cannot choose Disabled.

Deliver Conference Bridge Identifier

Check this check box for 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.

Early Offer support for voice and video calls (insert MTP if needed)

Check this check box if 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.

Because E2E RSVP provides an early offer by including an SDP in the initial INVITE, the early offer and E2E RSVP features are mutually exclusive on the SIP Profile Configuration window. When you choose E2E from the RSVP Over SIP drop-down list box, the Early Offer support for voice and video calls (insert MTP if needed) check box gets disabled.

Note When checked, the Media Termination Required check box on the Trunk Configuration window overrides the early offer configuration on the associated SIP profile. The Cisco Unified Communications Manager sends the MTP IP address and port with a single codec in the SDP in the initial INVITE.

Send send-receive SDP in mid-all INVITE

Check this check box to prevent Cisco Unified Communications Manager from sending an INVITE a=inactive SDP message during call hold or media break during supplementary services.

Note This check box applies only to early offer enabled SIP trunks and has no impact on SIP line calls.

When you enable Send send-receive SDP in mid-call INVITE for an early offer SIP trunk in tandem mode, Cisco 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, Cisco Unified Communications Manager 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 Cisco 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 To prevent the SDP mode from being set to inactive in a multiple-hold scenario, set the Duplex Streaming Enabled clusterwide service parameter (System > Service Parameters) to True.

SIP OPTIONS Ping

Enable OPTIONS Ping to monitor destination status for Trunks with service type "None (Default)"

Check this check box if you want 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, Cisco Unified Communications Manager tries to reroute the calls by using other trunks or by using a different address.

If this check box is not checked, the SIP trunk does not track the status of SIP trunk destinations.

When this check box is checked, you can configure two request timers.

Ping Interval for In-service and Partially In-service Trunks (seconds)

This field 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.

Ping Interval for Out-of-service SIP Trunks (seconds)

This field 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.

Ping Retry Timer (milliseconds)

This field specifies the maximum waiting time before retransmitting the OPTIONS request.

Valid values range from 100 to 1000 milliseconds. The default value specifies 500 milliseconds.

Ping Retry Count

This field specifies the number of times that Cisco 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.


Additional Information

See the "Related Topics" section.

Synchronizing a SIP Profile With Affected SIP Devices

To synchronize SIP devices with a SIP profile that has undergone configuration changes, perform the following procedure, which applies any outstanding configuration settings in the least-intrusive manner possible. (For example, a reset/restart may not be required on some affected devices.)

Procedure


Step 1 Choose Device > Device Settings > SIP Profile.

The Find and List SIP Profiles window displays.

Step 2 Choose the search criteria to use.

Step 3 Click Find.

The window displays a list of SIP Profiles that match the search criteria.

Step 4 Click the SIP profile to which you want to synchronize applicable SIP devices. The SIP Profile Configuration window displays.

Step 5 Make any additional configuration changes.

Step 6 Click Save.

Step 7 Click Apply Config.

The Apply Configuration Information dialog displays.

Step 8 Click OK.


Additional Information

See the "Related Topics" section.

Related Topics

SIP Profile Configuration

SIP Profile Configuration Settings

Synchronizing a SIP Profile With Affected SIP Devices

Phone Configuration Settings, page 67-3

Trunk Configuration Settings, page 68-1

Resource Priority Namespace Network Domain Configuration Settings, page 18-1

Resource Priority Namespace List Configuration Settings, page 19-1

Understanding Session Initiation Protocol, Cisco Unified Communications Manager System Guide