SIP Profile Field Descriptions ------------------------------ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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 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 | 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{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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 | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | 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. | +----------------+-------------------------------------------------------------+ | 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. | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | 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. | +----------------+-------------------------------------------------------------+ Table: SDP Information Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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 | +----------------+-------------------------------------------------------------+ Table: Parameters used in Phone Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-----------------------------------------------------------+ | 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 | +----------------+-----------------------------------------------------------+ | 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 | +----------------+-----------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | 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 | +----------------+-------------------------------------------------------------+ | 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. | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | 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 | +----------------+------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | 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. | +----------------+------------------------------------------------------------+ | 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. | +----------------+------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+------------------------------------------------------------+ | Option | Description | +================+============================================================+ | 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. | +----------------+------------------------------------------------------------+ Table: Normalization Script Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +---------------+-----------------------------------------------------------+ | 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. | +---------------+-----------------------------------------------------------+ Table: Incoming Requests FROM URI Settings Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +--------------+----------------------------------------------------------+ | 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. | +--------------+----------------------------------------------------------+ Table: Trunk Specific Configuration Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+------------------------------------------------------------+ | 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{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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. | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | 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 | | +----------------+-------------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-------------------------------------------------------------+ | Option | Description | +================+=============================================================+ | 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. | +----------------+-------------------------------------------------------------+ Table: Trunk SIP OPTIONS Ping Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-----------------------------------------------------------+ | 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 | +----------------+-----------------------------------------------------------+ Table: Trunk SDP Information Tab .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-----------------------------------------------------------+ | 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. | +----------------+-----------------------------------------------------------+ .. tabularcolumns:: |p{4cm}|p{11cm}| +----------------+-----------------------------------------------------------------+ | Option | Description | +================+=================================================================+ | 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. | +----------------+-----------------------------------------------------------------+ | 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. | +----------------+-----------------------------------------------------------------+