.. rst-class:: chapter-with-expand Configure SIP Profiles ----------------------- 1. Log in as provider, reseller, or customer administrator. 2. Make sure that the hierarchy path is set to the node where the Cisco Unified Communications Manager is configured. 3. Perform one of the following: * If you signed in as a provider or reseller administrator, choose **Device Management > CUCM > SIP Profiles**. * If you signed in as a customer administrator, choose **Device Management > Advanced > SIP Profiles**. 4. Perform one of the following: * To add a new SIP profile, click **Add**, then go to Step 5. * To edit an existing SIP profile, choose the SIP profile to be updated by clicking it in the list of SIP profiles. Go to Step 6. 5. If the **Network Device List** popup window appears, select the NDL for the SIP profile from the drop-down menu. The window appears when you are on a nonsite hierarchy node. If you are at a site hierarchy node, the NDL associated with the site is automatically used. Note: The **Network Device List** drop-down menu only appears when a SIP profile is added; it does not appear when you edit a SIP profile. 6. Enter a unique name for the new SIP profile in the **Name** field, or modify the existing **Name** if desired. 7. On the **SIP Profile Information** tab, complete at minimum, the mandatory :ref:`sip_profile_information_fields`. 8. On the **SDP Information** tab, complete at minimum, the mandatory :ref:`sdp_information_fields`. 9. On the **Parameters used in Phone** tab, complete the required :ref:`parameters_used_in_phone_fields`. 10. On the **Normalization Script** tab, complete the required :ref:`normalization_script_fields`. 11. On the **Incoming Requests FROM URI Strings** tab, complete the required :ref:`incoming_requests_from_uri_strings_fields`. 12. On the **Trunk Specific Configuration** tab, complete at minimum, the mandatory :ref:`trunk_specific_configuration_fields`. 13. On the **Trunk SIP OPTIONS Ping** tab, complete the required :ref:`trunk_sip_options_ping_fields`. 14. On the **Trunk SDP Information** tab, complete the required :ref:`trunk_sdp_information_fields`. 15. Click **Save** to save a new SIP profile or to update an existing SIP profile. .. _sip_profile_information_fields: SIP Profile Information Fields .............................. .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | Name | Enter a name to identify the SIP profile; for example, | | (Mandatory) | 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 | | (Optional) | example, SIP for 8865. 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 | This field specifies the default payload type for RFC2833 | | Telephony | telephony event. See RFC 2833 for more information. | | Event Payload | Usually, the default value specifies the appropriate | | Type | payload type. Be sure that you have a good understanding | | (Optional) | of this parameter before changing it, as changes could | | | result in DTMF tones not being received or generated. | | | | | | Default-101 | | | | | | Range-96 to 127 | | | | | | This parameter's value affects calls with the following | | | conditions: | | | | | | - 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 | +----------------+-------------------------------------------------------------+ | Early Offer | This feature supports both standards-based G.Clear | | for G.Clear | (CLEARMODE) and proprietary Cisco Session Description | | Calls | Protocols (SDP). | | (Optional) | | | | To enable or disable Early Offer for G.Clear Calls, | | | choose one of the following options: | | | | | | - Disabled | | | - CLEARMODE | | | - CCD | | | - G.nX64 | | | - X-CCD | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | User-Agent and | This feature indicates how Unified CM handles the | | Server header | User-Agent and Server header information in a SIP | | information | message. | | (Mandatory) | | | | Choose one of the following options: | | | | | | - **Send Unified CM 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 CM | | | passes any contact headers through untouched. | | | - **Pass Through Received Information as Contact Header | | | Parameters** - If selected, the User-Agent and Server | | | header information is passed as Contact header | | | parameters. The User-Agent and Server header is | | | derived from the received Contact header parameters, | | | if present. Otherwise, they are taken from the | | | received User-Agent and Server headers. | | | - **Pass Through Received Information as User-Agent and | | | Server Header** - If selected, the User-Agent and Server | | | header information is passed as User-Agent and Server | | | headers. The User-Agent and Server header is derived | | | from the received Contact header parameters, if | | | present. Otherwise, they are taken from the received | | | User-Agent and Server headers. | | | | | | Default: Send Unified CM Version Information as | | | User-Agent Header | +----------------+-------------------------------------------------------------+ | Version in | This field specifies the portion of the installed build | | User Agent and | version that is used as the value of the User Agent and | | Server Header | Server Header in SIP requests. Possible values are: | | (Mandatory) | | | | - **Major and Minor**; for example, Cisco-CUCM10.6 | | | - **Major:** for example, Cisco-CUCM10 | | | - **Major, Minor and Revision**; for example, | | | Cisco-CUCM10.6.2 | | | - **Full Build**; for example, Cisco-CUCM10.6.2.98000-19 | | | - **None**; header is omitted | | | | | | Default: Major and Minor | +----------------+-------------------------------------------------------------+ | Dial String | Possible values are: | | Interpretation | | | (Mandatory) | - Phone number consists of characters 0-9, \*, #, and + | | | (others treated as URI addresses). This is the default | | | value. | | | - Phone number consists of characters 0-9, A-D, \*, #, | | | and + (others treated as URI addresses) | | | - Always treat all dial strings as URI addresses | +----------------+-------------------------------------------------------------+ | Redirect by | If you select this check box and configure this SIP Profile | | Application | on the SIP trunk, the Unified CM administrator can: | | (Optional) | | | | - 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 calls get routed correctly. | | | - Prevent a 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 stack level causes the call to be routed, not blocked. | | | This behavior occurs if you leave the **Redirect by | | | Application** check box clear. | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | Disable Early | By default, Unified CM signals the calling phone to play | | Media on 180 | local ringback if SDP is not received in the 180 or 183 | | (Optional) | response. If SDP is included in these responses, instead | | | of playing ringback locally, Unified CM connects media. | | | The calling phone then plays whatever the called device | | | is sending (such as ringback or busy signal). If you | | | receive no ringback, the device you are connecting to may | | | include SDP in the 180 response, but not send media | | | before 200OK response. In this case, select 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. | +----------------+-------------------------------------------------------------+ | Outgoing T.38 | The parameter allows the system to accept a signal from | | INVITE include | Microsoft Exchange that causes it to switch the call from | | audio mline | audio to T.38 fax. To use this feature, configure a SIP | | (Optional) | trunk with this SIP profile. | | | | | | Note: | | | | | | The parameter applies to SIP trunks only, not phones | | | that are running SIP or other endpoints. | +----------------+-------------------------------------------------------------+ | Use Fully | This feature enables Unified CM to relay a caller's | | Qualified | alphanumeric hostname by passing it to the called device | | Domain Name in | or outbound trunk as SIP header information. Enter one of | | SIP Requests | the following: | | (Optional) | | | | **f** - To disable this option. The IP address for Unified | | | CM is passed to the line device or outbound trunk instead | | | of the user’s hostname. | | | | | | **t** - To enable this option. Unified CM relays 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 originates from a line device on the Unified CM | | | cluster, and is routed on a SIP trunk, then the | | | configured Organizational Top-Level Domain (for example, | | | Cisco.com) is used in the Identity headers, such as From, | | | Remote-Party-ID, and P-Asserted-ID. If the call | | | originates from a trunk on Unified CM 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 | | | preserves 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 is used in the | | | Identity headers, such as From, Remote-Party-ID, and | | | P-Asserted-ID. | | | | | | Default: f - Disabled | +----------------+-------------------------------------------------------------+ | Assured | Select this check box for third-party AS-SIP endpoints and | | Services SIP | AS-SIP trunks to ensure proper Assured Service behavior. | | conformance | This setting provides specific Assured Service behavior | | (Optional) | that affects services such as Conference factory and | | | SRTP. | +----------------+-------------------------------------------------------------+ .. _sdp_information_fields: SDP Information Fields ...................... .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | SDP | Displays the SDP Transparency Profile Setting (read-only) | | Transparency | | | Profile | | | (Optional) | | +----------------+-------------------------------------------------------------+ | Accept Audio | Choose one of the following options: | | Codec | | | Preferences in | - **On** - Enables Unified CM to honor the preference of | | Received Offer | audio codecs in the received offer and preserve it | | (Optional) | while processing. | | | - **Off** - Enables Unified CM to ignore the preference of | | | audio codecs in a received offer and apply the locally | | | configured Audio Codec Preference List. The default | | | selects the service parameter configuration. | | | - **Default** - Selects the service parameter | | | configuration. | | | | | | Default: Default | +----------------+-------------------------------------------------------------+ | Require SDP | This feature determines how Unified CM handles midcall | | Inactive | updates to codecs or connection information such as IP | | Exchange for | address or port numbers. | | Mid-Call Media | | | Change | If you select this check box, during midcall codec or | | (Optional) | connection updates Unified CM 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, the Send | | | send-receive SDP in midcall INVITE parameter | | | overrides this parameter. | | | | | | If this check box is clear, Unified CM passes the midcall | | | SDP to the peer leg without sending a prior Inactive SDP | | | to break the media exchange. | | | | | | Default: Clear | +----------------+-------------------------------------------------------------+ | Allow RR/RS | Specifies the RR (RTDP bandwidth allocated to other | | bandwidth | participants in an RTP session) and RS (RTCP bandwidth | | modifier (RFC | allocated to active data senders) in RFC 3556. Options | | 3556) | are: | | (Mandatory) | | | | - Transport Independent Application Specific bandwidth | | | modifier (TIAS) and AS | | | - TIAS only | | | - AS only | | | - CT only | | | | | | Default: TIAS and AS | +----------------+-------------------------------------------------------------+ .. _parameters_used_in_phone_fields: Parameters used in Phone Fields ............................... .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-----------------------------------------------------------+ | Option | Description | +================+===========================================================+ | Timer Invite | This field specifies the time, in seconds, after which a | | Expires | SIP INVITE expires. The Expires header uses this value. | | (seconds) | | | (Optional) | Valid values: Any positive number | | | | | | Default: 180 seconds | +----------------+-----------------------------------------------------------+ | Timer Register | This field is intended to be used by SIP endpoints only. | | Delta | The endpoint receives this value through a TFTP config | | (seconds) | file. The endpoint reregisters Timer Register Delta | | (Optional) | seconds before the registration period ends. The | | | registration period gets determined by the value of the | | | ``SIP Station KeepAlive Interval`` | | | service parameter. | | | | | | Valid values: 0 to 32767 | | | | | | Default: 5 seconds | +----------------+-----------------------------------------------------------+ | Timer Register | This field is intended to be used by SIP endpoints only. | | Expires | The SIP endpoint receives the value through a TFTP config | | (seconds) | file. This field specifies the value that the phone that | | (Optional) | 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. | | | | | | Valid values: Any positive number | | | | | | Default: 3600 seconds (1 hour) | | | | | | If the endpoint sends a shorter Expires value than the | | | ``SIP Station Keepalive Interval`` service parameter, | | | Unified CM responds with a 423 "Interval Too Brief." | | | | | | If the endpoint sends a greater Expires value than the | | | ``SIP Station Keepalive Interval`` service parameter, | | | Unified CM responds with a 200 OK with the Keepalive | | | Interval value for Expires. | | | | | | Note: | | | | | | For mobile phones running SIP, Unified CM uses this value | | | instead of the ``SIP Station KeepAlive Interval`` service | | | parameter to determine the registration period. | | | | | | Note: | | | | | | 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 | This field specifies the lowest value, in milliseconds, | | (msec) | of the retransmission timer for SIP messages. | | (Optional) | | | | Valid values: Any positive number | | | | | | Default: 500 msec | +----------------+-----------------------------------------------------------+ | Timer T2 | This field specifies the highest value, in milliseconds, | | (msec) | of the retransmission timer for SIP messages. | | (Optional) | | | | Valid values: Any positive number | | | | | | Default: 4000 msec | +----------------+-----------------------------------------------------------+ | Retry INVITE | This field specifies the maximum number of times that an | | (Optional) | INVITE request gets retransmitted. | | | | | | Valid values: Any positive number | | | | | | Default: 6 | +----------------+-----------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | Retry | This field specifies the maximum number of times that a | | Non-INVITE | SIP message other than an INVITE request gets | | (Optional) | retransmitted. | | | | | | Valid values: Any positive number | | | | | | Default: 10 | +----------------+-------------------------------------------------------------+ | Start Media | This field designates the start real-time protocol (RTP) | | Port | port for media. | | (Optional) | | | | Range: 2048 to 65535 | | | | | | Default: 16384 | +----------------+-------------------------------------------------------------+ | Stop Media | This field designates the stop real-time protocol (RTP) | | Port | port for media. | | (Optional) | | | | Range: 2048 to 65535 | | | | | | Default: 32766 | +----------------+-------------------------------------------------------------+ | Call Pickup | This URI provides a unique address that the phone that is | | URI (Optional) | running SIP sends to Unified CM to invoke the call pickup | | | feature. | +----------------+-------------------------------------------------------------+ | Call Pickup | This URI provides a unique address that the phone that is | | Group URI | running SIP sends to Unified CM to invoke the call pickup | | (Optional) | group feature. | +----------------+-------------------------------------------------------------+ | Meet Me | This URI provides a unique address that the phone that is | | Service URI | running SIP sends to Unified CM to invoke the meet me | | (Optional) | conference feature. | +----------------+-------------------------------------------------------------+ | User Info | This field configures the user- parameter in the REGISTER | | (Optional) | message. Valid values are: | | | | | | - **None** - No value is inserted | | | - **Phone** - The value user-phone is inserted in the To, | | | From, and Contact Header for REGISTER | | | - **IP** - The value user-ip is inserted in the To, From, | | | and Contact Header for REGISTER | | | | | | Default: None | +----------------+-------------------------------------------------------------+ | DTMF DB Level | This field specifies the in-band DTMF digit tone level. | | (Optional) | Valid values are: | | | | | | - 6 dB below nominal | | | - 3 dB below nominal | | | - Nominal | | | - 3 dB above nominal | | | - 6 dB above nominal | | | | | | Default: Nominal | +----------------+-------------------------------------------------------------+ | Call Hold Ring | This parameter causes the phone to ring in cases where | | Back | you have another party on hold when you hang up a call. | | (Optional) | Valid values are: | | | | | | - **Off** - Off permanently and cannot be turned on and | | | off locally by the user interface | | | - **On** - On permanently and cannot be turned on and off | | | locally by the user interface | +----------------+-------------------------------------------------------------+ | Anonymous Call | The field configures anonymous call block. Valid values | | Block | are: | | (Optional) | | | | - **Off** - Disabled permanently and cannot be turned on | | | and off locally by the user interface | | | - **On** - Enabled permanently and cannot be turned on and | | | off locally by the user interface | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | Caller ID | This field configures caller ID blocking. When blocking | | Blocking | is enabled, the phone blocks its own number or email | | (Optional) | address from phones that have caller identification | | | enabled. Valid values are: | | | | | | - **Off** - Disabled permanently and cannot be turned on | | | and off locally by the user interface | | | - **On** - Enabled permanently and cannot be turned on and | | | off locally by the user interface | +----------------+------------------------------------------------------------+ | Do Not Disturb | This field sets the Do Not Disturb (DND) feature. Valid | | Control | values are: | | (Optional) | | | | - **User** - The dndControl parameter for the phone | | | specifies 0. | | | - **Admin** - The dndControl parameter for the phone | | | specifies 2. | +----------------+------------------------------------------------------------+ | Telnet Level | Cisco Unified IP Phones 7940 and 7960 do not support SSH | | for 7940 and | for sign-in access or HTTP that is used to collect logs. | | 7960 | However, these phones support Telnet, which lets the user | | (Optional) | 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 | +----------------+------------------------------------------------------------+ | Resource | This field enables the administrator to select one of the | | Priority | cluster's defined Resource Priority Namespace network | | Namespace | domains for assignment to a line using its SIP Profile. | | (Optional) | | +----------------+------------------------------------------------------------+ | Timer Keep | Unified CM requires a keepalive mechanism to support | | Alive Expires | redundancy. This field specifies the interval between | | (seconds) | keepalive messages sent to the backup Unified CM to | | (Optional) | ensure its availability for failover. | | | | | | Default: 120 seconds | +----------------+------------------------------------------------------------+ | Timer | This field specifies the time, in seconds, after which a | | Subscribe | subscription expires. This value gets inserted into | | Expires | the\ `` Expires`` header field. | | (seconds) | | | (Optional) | Valid values: Any positive number | | | | | | Default: 120 seconds | +----------------+------------------------------------------------------------+ | Timer | Use this parameter with the ``Timer Subscribe Expires`` | | Subscribe | setting. The phone resubscribes Timer Subscribe Delta | | Delta | seconds before the subscription period ends, as governed | | (seconds) | by ``Timer Subscribe Expires``. | | (Optional) | | | | Range: 3 to 15 seconds | | | | | | Default: 5 seconds | +----------------+------------------------------------------------------------+ | Maximum | Use this configuration variable to determine the maximum | | Redirections | number of times that the phone allows a call to be | | (Optional) | redirected before dropping the call. | | | | | | Default: 70 redirections | +----------------+------------------------------------------------------------+ | Off hook To | This field specifies the time in microseconds that passes | | First Digit | when the phone goes off hook and the first digit timer | | Timer (msec) | gets set. | | (Optional) | | | | Range: 0 to 15,000 microseconds | | | | | | Default: 15,000 microseconds | +----------------+------------------------------------------------------------+ | Call Forward | This URI provides a unique address that the phone that is | | URI (Optional) | running SIP sends to Unified CM to invoke the call | | | forward feature. | +----------------+------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | Speed Dial | This URI provides a unique address that the phone that is | | (Abbreviated | running SIP sends to Unified CM to invoke the abbreviated | | Dial) URI | dial feature. | | (Optional) | | | | 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. | | | Unified CM translates the abbreviated dial digits into | | | the configured digit string and extends 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 | Select this check box to join the remaining conference | | Join Enabled | participants when a conference initiator using a Cisco | | (Optional) | Unified IP Phone 7940 or 7960 hangs up. Leave it | | | clear if you do not want to join the remaining | | | conference participants. | | | | | | 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 | Select this check box to enable setting connection address | | (Optional) | to 0.0.0.0 per RFC2543 when call hold is signaled to | | | Unified CM. This allows backward compatibility with | | | endpoints that do not support RFC3264. | +----------------+------------------------------------------------------------+ | Semi Attended | This check box determines whether the Cisco Unified IP | | Transfer | Phones 7940 and 7960 caller can transfer an attended | | (Optional) | transfer's second leg while the call is ringing. Select | | | the check box if you want semi attended transfer enabled; | | | leave it clear 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 | Select this check box if you want voice activation | | (Optional) | detection (VAD) enabled; leave it clear if you want VAD | | | disabled. When VAD is enabled, no media is sent when voice | | | is detected. | +----------------+------------------------------------------------------------+ | Stutter | Select this check box if you want stutter dial tone when | | Message | the phone goes off hook and a message is waiting. Leave | | Waiting | clear if you do not want a stutter dial tone when a | | (Optional) | message is waiting. | | | | | | This setting supports Cisco Unified IP Phones 7960 and | | | 7940 that run SIP. | +----------------+------------------------------------------------------------+ | MLPP User | Select this check box to enable MLPP User Authorization. | | Authorization | MLPP User Authorization requires the phone to send in an | | (Optional) | MLPP username and password. | +----------------+------------------------------------------------------------+ .. _normalization_script_fields: Normalization Script Fields ........................... .. tabularcolumns:: |p{3.5cm}|p{12cm}| +---------------+-----------------------------------------------------------+ | Option | Description | +===============+===========================================================+ | Normalization | From the drop-down list, choose the script that you | | Script | want to apply to this SIP profile. | | | | | | To import another script from Unified CM, go to the SIP | | | Normalization Configuration window (Device Device | | | Settings SIP Normalization Script), and import a new | | | script. | +---------------+-----------------------------------------------------------+ | Enable Trace | Select this check box to enable tracing within the script | | | or clear this check box to disable tracing. When selected,| | | 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 | | | is not recommended under normal operating conditions. | +---------------+-----------------------------------------------------------+ | Script | Enter parameter names and parameter values in the | | Parameters | **Script Parameters** box as comma-delineated key-value | | | pairs. Valid values include all characters except equals | | | signs (-), semicolons (;), and nonprintable characters, | | | such as tabs. You can enter a parameter name with no | | | value. | | | | | | Alternatively, to add another parameter line from Unified | | | CM, click the + (plus) button. To delete a parameter | | | line, click the - (minus) button. | +---------------+-----------------------------------------------------------+ .. _incoming_requests_from_uri_strings_fields: Incoming Requests FROM URI Strings Fields ......................................... .. tabularcolumns:: |p{3.5cm}|p{12cm}| +--------------+----------------------------------------------------------+ | Option | Description | +==============+==========================================================+ | Caller ID DN | 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 | Enter a caller name to override the caller name that is | | | received from the originating SIP Device. | +--------------+----------------------------------------------------------+ .. _trunk_specific_configuration_fields: Trunk Specific Configuration Fields ................................... .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | Reroute | Unified CM only accepts calls from a SIP device whose IP | | Incoming | address matches the destination address of the configured | | Request to new | SIP trunk. In addition, the port on which the SIP message | | Trunk based on | arrives must match the one that is configured on the SIP | | | trunk. After Unified CM accepts the call, Unified CM uses | | | the configuration for this setting to determine whether | | | to reroute the call to another trunk. | | | | | | From the drop-down list, choose the method that | | | Unified CM 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. Unified | | | CM, which identifies the trunk by the incoming | | | packet's source IP address 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. Unified CM parses the IP | | | address or domain name and the signaling port number | | | in the incoming request's header. Unified CM then | | | reroutes the call to the SIP trunk using that IP | | | address and port. If no SIP trunk is identified, the | | | call occurs on the trunk where 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, Unified CM | | | performs the following: | | | | | | - parses the Call-Info header | | | - looks for the parameter ``purpose-x-cisco-origIP`` | | | - uses the IP address or domain name and signaling | | | port number in the header to reroute the call to | | | the SIP trunk using the IP address and port | | | | | | If the parameter is not in the header, or no SIP trunk | | | is identified, the call occurs on the SIP trunk where | | | the call arrived. | | | | | | Default: Never | | | | | | Note: | | | | | | This setting does not work for SIP trunks connected to: | | | | | | - A Unified CM IM and Presence Service proxy server. | | | - Originating gateways in different Unified CM groups | +----------------+------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | RSVP Over SIP | This field configures RSVP over SIP trunks. From the | | | drop-down list, choose the method that Unified CM | | | uses to configure RSVP over SIP trunks: | | | | | | - **Local RSVP** - In a local configuration, RSVP occurs | | | within each cluster, between the endpoint 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 endpoints, | | | including within the local cluster and over the WAN. | +----------------+-------------------------------------------------------------+ | Resource | Select a configured Resource Priority Namespace list from | | Priority | the drop-down menu. The Namespace List is configured in | | Namespace List | Unified CM in the Resource Priority Namespace List menu. | | | You can access the menu in Unified CM from System MLPP > | | | Namespace. | +----------------+-------------------------------------------------------------+ | Fall back to | Select this check box if you want to allow failed end-to-end| | local RSVP | RSVP calls to fall back to local RSVP to establish the | | | call. If this check box is clear, end-to-end RSVP calls that| | | cannot establish an end-to-end connection fail. | +----------------+-------------------------------------------------------------+ | SIP Rel1XX | This field configures SIP Rel1XX, which determines | | Options | whether all SIP provisional responses (other than 100 | | | Trying messages) are sent reliably to the remote SIP | | | endpoint. Valid values are: | | | | | | - **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. | +----------------+-------------------------------------------------------------+ | Video Call | Video Call Traffic Class determines the type of video | | Traffic Class | endpoint or trunk that the SIP Profile is associated | | | with. From the drop-down list, select one of: | | | | | | - **Immersive** - High-definition immersive video. | | | - **Desktop** - Standard desktop video. | | | - **Mixed** - A mix of immersive and desktop video. | | | | | | Unified CM Locations Call Admission Control (CAC) | | | reserves bandwidth from two Locations video bandwidth | | | pools, Video Bandwidth and Immersive Bandwidth. The pool | | | used depends on the type of call determined by the Video | | | Call Traffic Class. Refer to the “Call Admission Control” | | | chapter of the Cisco Unified Communications Manager | | | System Guide for more information. | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | Calling Line | Select one of: | | Identification | | | Presentation | - **Strict From URI presentation Only** - To select the | | (Mandatory) | network-provided identity | | | - **Strict Identity Headers presentation Only** - To | | | select the user-provided identity | | | - **Default** - To select the system default calling line | | | identification | | | | | | Default: Default | +----------------+-------------------------------------------------------------+ | Session | Session Timer with Update: The session refresh timer | | Refresh Method | allows for periodic refresh of SIP sessions. This allows | | (Mandatory) | the Unified CM and remote agents to determine whether the | | | SIP session is still active. Prior to Release 10.01, when | | | the Unified CM received a refresh command, it supported | | | receiving either Invite or Update SIP requests to refresh | | | the session. When the Unified CM 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 CM can send both | | | Update and Invite requests. | | | | | | Specify whether to use **Invite** or **Update** as the | | | Session Refresh Method. | | | | | | Default: Invite | | | | | | Note: | | | | | | Sending a midcall Invite request requires specifying | | | an offer SDP in the request. This means that the far | | | end must send an answer SDP in the Invite response. | | | | | | Update: Unified CM requests a SIP Update if the SIP | | | session's far end supports the Update method in the | | | Supported or Require headers. When sending the Update | | | request, the Unified CM includes an SDP. This | | | simplifies the session refresh since no SDP offer or | | | answer exchange is required. | | | | | | Note: | | | | | | If the far end of the SIP session does not support | | | the Update method, the Unified CM continues using the | | | Invite method for session refresh. | +----------------+-------------------------------------------------------------+ | Early Offer | This field configures Early Offer support for voice and | | Support for | video calls. When enabled, Early Offer support includes a | | voice and | session description in the initial INVITE for outbound | | video calls | calls. Early Offer configuration settings on SIP profile | | (Mandatory) | 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 | | | CM sends the MTP IP address and port with a single codec in | | | the SDP in the initial INVITE. | | | | | | From the drop-down list box, select one of the following | | | three options: | | | | | | * **Disabled (Default value)** - Disables Early Offer; | | | no SDP will be included in the initial INVITE for | | | outbound calls. | | | * **Best Effort (no MTP Inserted)** | | | | | | * Provide Early Offer for the outbound call only when | | | caller side's media port, IP and codec information is | | | available. | | | * 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. | | | * **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. | | | | | | Default: Disabled (Default value) | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | Enable ANAT | This option allows a dual-stack SIP trunk to offer both | | | IPv4 and IPv6 media. | | | | | | Selecting the **Enable ANAT** and **MTP Required** check | | | boxes sets Unified CM to insert a dual-stack MTP and send an| | | offer with two m-lines, for IPv4 and IPv6. If a dual- stack | | | MTP cannot be allocated, Unified CM sends an INVITE without | | | SDP. | | | | | | When you select the **Enable ANAT** check box and the | | | **Media Termination Point Required** check box is clear, | | | Unified CM sends an INVITE without SDP. | | | | | | When the **Enable ANAT** and **MTP Required** check boxes | | | are cleared (or when an MTP cannot be allocated), Unified | | | CM sends an INVITE without SDP. | | | | | | When you clear the **Enable ANAT** check box but you | | | select the **MPT Required** check box, consider the | | | information, which assumes that an MTP can be allocated: | | | | | | - Unified CM sends an IPv4 address in the SDP for SIP | | | trunks with an IP Addressing Mode of IPv4 Only. | | | - Unified CM sends an IPv6 address in the SDP for SIP | | | trunks with an IP Addressing Mode of IPv6 Only. | | | - For dual-stack SIP trunks, Unified CM determines which | | | IP address type to send in the SDP based on the | | | configuration for the IP Addressing Mode Preference | | | for Media enterprise parameter. | +----------------+-------------------------------------------------------------+ | Deliver | When checked, the SIP trunk passes the b-number | | Conference | identifying the conference bridge across the trunk | | Bridge | instead of changing the b-number to the null value. | | Identifier | | | | The terminating side does not require this field. | | | | | | Selecting this check box is not required for Open | | | Recording Architecture (ORA) SIP header enhancements to | | | the Recording feature to work. | | | | | | Selecting this check box allows the recorder to coordinate | | | recording sessions where the parties are participating in | | | a conference. | +----------------+-------------------------------------------------------------+ | Allow | Select this check box to allow passthrough of configured | | Passthrough of | line device caller information from the SIP trunk. | | Configured | | | Line Device | | | Caller | | | Information | | +----------------+-------------------------------------------------------------+ | Reject | Select this check box to reject anonymous incoming calls. | | Anonymous | | | Incoming Calls | | +----------------+-------------------------------------------------------------+ | Reject | Select this check box to reject anonymous outgoing calls. | | Anonymous | | | Outgoing Calls | | +----------------+-------------------------------------------------------------+ | Send ILS | When this check box is selected, for calls routed to a | | Learned | learned directory URI, learned number, or learned pattern, | | Destination | Unified CM: | | Route String | | | | - adds the ``x-cisco-dest-route-string`` header to | | | outgoing SIP INVITE and SUBSCRIBE messages | | | - inserts the destination route string into the header | | | | | | When this check box is clear, Unified CM does not add the | | | x-cisco-dest-route-string header to any SIP messages. | | | | | | The x-cisco-dest-route-string header allows Unified CM to | | | route calls across a Session Border Controller. | +----------------+-------------------------------------------------------------+ .. _trunk_sip_options_ping_fields: Trunk SIP OPTIONS Ping Fields ............................. .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-----------------------------------------------------------+ | Option | Description | +================+===========================================================+ | Enable OPTIONS | Select this check box if you want to enable the SIP | | Ping to | OPTIONS feature. SIP OPTIONS are requests to the | | monitor | configured destination address on the SIP trunk. If the | | destination | remote SIP device is unresponsive or returns a SIP error | | status for | response such as 503 Service Unavailable or 408 Timeout, | | Trunks with | Unified CM reroutes the calls by using other trunks or a | | Service Type | different address. | | "None | | | (Default)" | If this check box is clear, the SIP trunk does not track | | | the status of SIP trunk destinations. | | | | | | When this check box is selected, you can configure two | | | request timers. | +----------------+-----------------------------------------------------------+ | Ping Interval | This field configures the time duration between SIP | | for In-service | OPTIONS requests when the remote peer is responding and | | and Partially | the trunk is marked as In Service. If at least one IP | | In-service | address is available, the trunk is In Service; if all IP | | Trunks | addresses are unavailable, the trunk is Out of Service. | | (seconds) | | | | Default: 60 seconds | | | | | | Range: 5 to 600 seconds | +----------------+-----------------------------------------------------------+ | Ping Interval | This field configures the time duration between SIP | | for | OPTIONS requests when the remote peer is not responding | | Out-of-service | and the trunk is marked as Out of Service. The remote | | Trunks | peer may be marked as Out of Service if: | | (seconds) | | | | - it fails to respond to OPTIONS | | | - it sends 503 or 408 responses | | | - 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. | | | | | | Default: 120 seconds | | | | | | Range: 5 to 600 seconds | +----------------+-----------------------------------------------------------+ | Ping Retry | This field specifies the maximum waiting time before | | Timer (msec) | retransmitting the OPTIONS request. | | | | | | Range: 100 to 1000 milliseconds | | | | | | Default: 500 milliseconds | +----------------+-----------------------------------------------------------+ | Ping Retry | This field specifies the number of times that Unified CM | | Count | 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. | | | | | | Range: 1 to 10 | | | | | | Default: 6 | +----------------+-----------------------------------------------------------+ .. _trunk_sdp_information_fields: Trunk SDP Information Fields ............................ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-----------------------------------------------------------------+ | Option | Description | +================+=================================================================+ | Send | Select this check box to prevent Unified CM from sending | | send-receive | an INVITE a-inactive SDP message during call hold or media | | SDP in midcall | break during supplementary services. | | INVITE | | | | 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 midcall INVITE | | | for an early offer SIP trunk in tandem mode, Unified CM | | | 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, Unified CM depends | | | on the SIP devices to reestablish media path by sending | | | either a delayed INVITE or midcall INVITE with send-recv | | | SDP. | | | | | | When you enable Send send-receive SDP in midcall INVITE | | | and Require SDP Inactive Exchange for Mid-Call Media | | | Change on the same SIP Profile, the Send send-receive SDP | | | in midcall INVITE overrides the Require SDP Inactive | | | Exchange for Mid-Call Media Change, so Unified CM does | | | not send an INVITE with a-inactive SDP in midcall 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 in Unified CM | | | (System Service Parameters) to True. | +----------------+-----------------------------------------------------------------+ | Allow | If the check box is selected, Unified CM allows supported SIP | | Presentation | endpoints to use the Binary Floor Control Protocol (BFCP) | | Sharing using | to enable presentation sharing. | | BFCP | | | | The use of BFCP creates an added 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 check box is clear, Unified CM rejects BFCP offers | | | from devices associated with the SIP profile. The BFCP | | | application line and associated media line ports are set | | | to 0 in the answering SDP message. | | | | | | Default: Clear | | | | | | Note: | | | | | | 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. | | | | | | For more information on BFCP, refer to the Cisco Unified | | | Communications Manager System Guide. | +----------------+-----------------------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +----------------+-----------------------------------------------------------------+ | Option | Description | +================+=================================================================+ | Allow iX | Select this check box to enable support for iX media | | Application | channel. | | Media | | +----------------+-----------------------------------------------------------------+ | Allow multiple | This option applies when incoming SIP signals do not | | codecs in | indicate support for multiple codec negotiation and | | answer SDP | Unified CM can finalize the negotiated codec. | | | | | | When this check box is selected, the endpoint behind the | | | trunk can handle multiple codecs in the answer SDP. | | | | | | For example, an endpoint that supports multiple codec | | | negotiation calls the SIP trunk, and Unified CM 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 CM identifies that the trunk can | | | handle multiple codec negotiation, and sends SIP response | | | messages to both endpoints with multiple common codecs. | | | | | | When clear, Unified CM identifies that the endpoint | | | behind the trunk cannot handle multiple codec | | | negotiation, unless SIP contact header URI states it can. | | | Unified CM continues the call with single codec | | | negotiation. | +----------------+-----------------------------------------------------------------+