Cisco UCM SIP Profile Configuration
Cisco UCM SIP Profile Configuration is a SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.
SIP Profile Information
Name
Enter a name to identify the SIP profile; for example, SIP_7962. The value can include 1 to 50 characters, including alphanumeric characters, dot, dash, and underscores.
Description
This field identifies the purpose of the SIP profile; for example, SIP for 7965.
Default MTP Telephony Event Payload Type
This field specifies the default payload type for RFC2833 telephony event. See RFC 2833 for more information. In most cases, the default value specifies the appropriate payload type. Be sure that you have a firm understanding of this parameter before changing it, as changes could result in DTMF tones not being received or generated. The default value specifies 101 with range from 96 to 127.
The value of this parameter affects calls with the following conditions:
- The call is an outgoing SIP call from Cisco Unified Communications Manager.
- For the calling SIP trunk, the Media Termination Point Required check box is checked on the SIP Trunk Configuration window.
Resource Priority Namespace List
Select a configured Resource Priority Namespace Network Domain list from the drop-down menu. Configure the lists in the Resource Priority Namespace Network Domain menu that is accessed from the System menu.
Redirect by Application
Checking this check box and configuring this SIP Profile on the SIP trunk allows the Cisco Unified Communications Manager administrator to
- Apply a specific calling search space to redirected contacts that are received in the 3xx response.
- Apply digit analysis to the redirected contacts to make sure that the call gets routed correctly.
- Prevent DOS attack by limiting the number of redirection (recursive redirection) that a service parameter can set.
- Allow other features to be invoked while the redirection is taking place.
Getting redirected to a restricted phone number (such as an international number) means that handling redirection at the stack level will cause the call to be routed instead of being blocked. This represents the behavior that you will get if the Redirect by Application check box is unchecked.
Disable Early Media on 180
By default, Cisco Unified Communications Manager will signal the calling phone to play local ringback if SDP is not received in the 180 or 183 response. If SDP is included in the 180 or 183 response, instead of playing ringback locally, Cisco Unified Communications Manager will connect media, and the calling phone will play whatever the called device is sending (such as ringback or busy signal). If you do not receive ringback, the device to which you are connecting may be including SDP in the 180 response, but it is not sending any media before the 200OK response. In this case, check this check box to play local ringback on the calling phone and connect the media upon receipt of the 200OK response
Note Even though the phone that is receiving ringback is the calling phone, you need the configuration on the called device profile because it determines the behavior.
Outgoing T.38 INVITE Include Audio mline
The parameter allows the system to accept a signal from Microsoft Exchange that causes it to switch the call from audio to T.38 fax. To use this feature, you must also configure a SIP trunk with this SIP profile.
Note The parameter applies to SIP trunks only, not phones that are running SIP or other endpoints.
Parameters used in Phone
Timer Invite Expires (seconds)
This field specifies the time, in seconds, after which a SIP INVITE expires. The Expires header uses this value. Valid values include any positive number; 180 specifies the default.
Timer Register Delta (seconds)
Use this parameter in conjunction with the Timer Register Expires setting. The phone will reregister Timer Register Delta seconds before the registration period ends. The registration period gets determined by the value of the SIP Station Keepalive Interval service parameter. Valid values range from 32767 to 0. Default specifies 5.
Timer Register Expires (seconds)
This field specifies the value that the phone that is running SIP will send in the Expires header of the REGISTER message. Valid values include any positive number; however, 3600 (1 hour) specifies the default value. In the 200OK response to REGISTER, Cisco Unified Communications Manager will include an Expires header with the configured value of the SIP Station KeepAlive Interval service parameter. This value in the 200OK determines the time, in seconds, after which the registration expires. The phone will refresh the registration Timer Register Delta seconds before the end of this interval.
Timer T1 (msec)
This field specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 500.
Timer T2 (msec)
This field specifies the highest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 4000.
Retry INVITE
This field specifies the maximum number of times that an INVITE request will be retransmitted. Valid values include any positive number. Default specifies 6.
Retry Non-INVITE
This field specifies the maximum number of times that a SIP message other than an INVITE request will be retransmitted. Valid values include any positive number. Default specifies 10.
Start Media Port
This field designates the start real-time protocol (RTP) port for media. Media port ranges from 16384 to 32766. Default specifies 16384.
Stop Media Port
This field designates the stop real-time protocol (RTP) port for media. Media port ranges from 16384 to 32766. Default specifies 32766.
Call Pickup URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup feature.
Call Pickup Group Other URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup group other feature.
Call Pickup Group URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup group feature.
Meet Me Service URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the meet me conference feature.
User Info
This field configures the user= parameter in the REGISTER message.
Valid values follow:
- none—No value gets inserted.
- •phone—The value user=phone gets inserted in the To, From, and Contact Headers for REGISTER.
- •ip—The value user=ip gets inserted in the To, From, and Contact Headers for REGISTER.
DTMF DB Level
This field specifies in-band DTMF digit tone level. Valid values follow:
- 1 to 6 dB below nominal
- 2 to 3 dB below nominal
- 3 nominal
- 4 to 3 dB above nominal
- 5 to 6 dB above nominal
Call Hold Ring Back
If you have a call on hold and are talking on another call, when you hang up the call, this parameter causes the phone to ring to let you know that you still have another party on hold. Valid values follow:
- Off permanently and cannot be turned on and off locally by using the user interface.
- On permanently and cannot be turned on and off locally by using the user interface.
Anonymous Call Block
This field configures anonymous call block. Valid values follow:
- Off—Disabled permanently and cannot be turned on and off locally by using the user interface.
- On—Enabled permanently and cannot be turned on and off locally by using the user interface.
Caller ID Blocking
This field configures caller ID blocking. When blocking is enabled, the phone blocks its own number or e-mail address from phones that have caller identification enabled. Valid values follow:
- Off—Disabled permanently and cannot be turned on and off locally by using the user interface.
- On—Enabled permanently and cannot be turned on and off locally by using the user interface.
Do Not Disturb Control
This field sets the Do Not Disturb (DND) feature. Valid values follow:
- User—The dndControl parameter for the phone should specify 0.
- Admin—The dndControl parameter for the phone should specify 2.
Telnet Level for 7940 and 7960
Cisco Unified IP Phones 7940 and 7960 do not support ssh for login access or HTTP that is used to collect logs; however, these phones support Telnet, which lets the user control the phone, collect debugs, and look at configuration settings. This field controls the telnet_level configuration parameter with the following possible values:
- Disabled (no access)
- Limited (some access but cannot run privileged commands)
- Enabled (full access)
Timer Keep Alive Expires (seconds)
Cisco Unified Communications Manager requires a keepalive mechanism to support redundancy. This field specifies the interval between keepalive messages that are sent to the backup Cisco Unified Communications Manager to ensure that it is available in the event that a failover is required.
Timer Subscribe Expires (seconds)
This field specifies the time, in seconds, after which a subscription expires. This value gets inserted into the Expires header field. Valid values include any positive number; however, 120 specifies the default value.
Timer Subscribe Delta (seconds)
Use this parameter in conjunction with the Timer Subscribe Expires setting. The phone will resubscribe Timer Subscribe Delta seconds before the subscription period ends, as governed by Timer Subscribe Expires. Valid values range from 3 to 15. Default specifies 5.
Maximum Redirections
Use this configuration variable to determine the maximum number of times that the phone will allow a call to be redirected before dropping the call. Default specifies 70 redirections.
Off Hook to First Digit Timer (microseconds)
This field specifies the time in microseconds that passes when the phone goes off hook and the first digit timer gets set. The value ranges from 0 - 15,000 microseconds. Default specifies 15,000 microseconds.
Call Forward URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call forward feature.
Abbreviated Dial URI
This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the abbreviated dial feature.
Speed dials that are not associated with a line key (abbreviated dial indices) will not download to the phone. The phone will use the feature indication mechanism (INVITE with Call-Info header) to indicate when an abbreviated dial number has been entered. The request URI will contain the abbreviated dial digits (for example, 14), and the Call-Info header will indicate the abbreviated dial feature. Cisco Unified Communications Manager will translate the abbreviated dial digits into the configured digit string and extend the call with that string. If no digit string has been configured for the abbreviated dial digits, a 404 Not Found response gets returned to the phone.
Conference Join Enabled
This check box determines whether the Cisco Unified IP Phones 7940 or 7960, when the conference initiator that is using that phone hangs up, should attempt to join the remaining conference attendees. Check the check box if you want to join the remaining conference attendees; leave it unchecked if you do not want to join the remaining conference attendees.
Note This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.
RFC 2543 Hold
Check this check box to enable setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Cisco Unified Communications Manager. This allows backward compatibility with endpoints that do not support RFC3264.
Semi Attended Transfer
This check box determines whether the Cisco Unified IP Phones 7940 and 7960 caller can transfer the second leg of an attended transfer while the call is ringing. Check the check box if you want semi-attended transfer enabled; leave it unchecked if you want semi-attended transfer disabled.
Note This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.
Enable VAD
Check this check box if you want voice activation detection (VAD) enabled; leave it unchecked if you want VAD disabled. When VAD is enabled, no media gets transmitted when voice is detected.
Stutter Message Waiting
Check this check box if you want stutter dial tone when the phone goes off hook and a message is waiting; leave unchecked if you do not want a stutter dial tone when a message is waiting.
This setting supports Cisco Unified IP Phones 7960 and 7940 that run SIP.
Call Stats
Check this check box if you want RTP statistics in BYE requests and responses enabled; leave unchecked if you want RTP statistics in BYE requests and responses disabled.
If this check box is checked, the phone inserts the headers RTP-RxStat and RTP-TxStat as follows:
- RTP-RxStat:Dur=a,Pkt=b,Oct=c,LatePkt=d,LostPkt=e,AvgJit=f
- RTP-TxStat: Dur=g,Pkt=h,Oct=i
where:
- Dur—Total number of seconds since the beginning of reception or transmission.
- Pkt—Total number of RTP packets that are received or transmitted.
- Oct—Total number of RTP payload octets that are received or transmitted (not including RTP header).
- LatePkt—Total number of late RTP packets that are received.
- LostPkt—Total number of lost RTP packets that are received (not including the late RTP packets).
- AvgJit—Average jitter, which is an estimate of the statistical variance of the RTP packet interarrival time, measured in timestamp unit and calculated according to RFC 1889.
- a, b, c, d, e, f, g, h, and i—Integers
Trunk Specific Configuration
Reroute Incoming Request to new Trunk based on
Cisco Unified Communications Manager only accepts calls from the SIP device whose IP address matches the destination address of the configured SIP trunk. In addition, the port on which the SIP message arrives must match the one that is configured on the SIP trunk. After Cisco Unified Communications Manager accepts the call, Cisco Unified Communications Manager uses the configuration for this setting to determine whether the call should get rerouted to another trunk.
From the drop-down list box, choose the method that Cisco Unified Communications Manager uses to identify the SIP trunk where the call gets rerouted:
- Never—If the SIP trunk matches the IP address of the originating device, choose this option, which equals the default setting. Cisco Unified Communications Manager, which identifies the trunk by using the source IP address of the incoming packet and the signaling port number, does not route the call to a different (new) SIP trunk. The call occurs on the SIP trunk on which the call arrived.
- Contact Info Header—If the SIP trunk uses a SIP proxy, choose this option. Cisco Unified Communications Manager parses the contact header in the incoming request and uses the IP address or domain name and signaling port number that is specified in the header to reroute the call to the SIP trunk that uses the IP address and port. If no SIP trunk is identified, the call occurs on the trunk on which the call arrived.
- Call-Info Header with purpose=x-cisco-origIP—If the SIP trunk uses a Customer Voice Portal (CVP) or a Back-to-Back User Agent (B2BUA), choose this option. When the incoming request is received, Cisco Unified Communications Manager parses the Call-Info header, looks for the parameter, purpose=x-cisco-origIP, and uses the IP address or domain name and the signaling port number that is specified in the header to reroute the call to the SIP trunk that uses the IP address and port. If the parameter does not exist in the header or no SIP trunk is identified, the call occurs on the SIP trunk on which the call arrived.
Data Communication Networks Open Systems Interconnection (OSI) Model and Notation, Service Definition (Recommendations X.200-X.219, Vol VIII Fascicle VIII.4)
Basic Setup:
