This SIP Peer Profile form is used to configure SIP trunks with the following:
the local account information
the outbound proxy server
the Calling Line ID information
the policies applicable to the SIP trunk
the authentication information
the Outgoing DID Ranges for CPN Substitution
RTP (voice) stream packet rate
SIP private networking trunks
Use this form when performing the following tasks:
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Parameter |
Description |
Default Value |
|
SIP Peer Profile Label |
Enter an alphanumeric string up to nine characters for the SIP Peer Profile. |
Blank |
|
Network Element |
Select the appropriate Network Element name (programmed in the Network Elements form) from the pull-down list. |
Blank |
|
Local Account Information |
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Registration User Name |
Enter the DIDs assigned by the Service Provider if registration is required. The default is blank. Enter one or more user names. The field accepts a maximum of 60 characters. The maximum number of alphanumeric characters allowed per user name is 26. If using a range, you must use only digits 0 through 9 with a dash (-) separating the first number in the range and the last. You can enter a mix of numerical ranges and single usernames (for example, "6135554000-6135554400, 6135554500"). Use a comma to separate user names and ranges. The first and last characters cannot be a comma or a dash. For Service Providers and SIP Services that require the 3300 to register, this field is used to indicate the user names that they wish us to register with. Normally (when required) this is a single user name but it could also refer to a range of DIDs being supplied by the service provider. |
Blank |
|
Address Type |
Select the address type for the local host. Two types are available:
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FQDN |
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Call Routing and Administration Options |
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Interconnect Restriction |
Enter the Interconnect Restriction number that is used to restrict device interconnections. |
1 |
|
Maximum Simultaneous Calls |
Enter the maximum allowable number of incoming and outgoing simultaneous calls for this peer.
|
400 |
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Outbound Proxy Server |
Select the network element to be used for the Outbound Proxy Server. |
|
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SMDR Tag |
Enter the SMDR tag number. The range is 1-9998. This tag number is used in SMDR logs for both incoming and outgoing calls when SMDR is enabled. |
Blank |
|
Trunk Service |
Enter the Trunk Service number where the COS/COR and incoming digit modification setting are set. |
1 |
|
Zone |
Use this field to group together devices to which compression policies can be applied. |
1 |
|
Alternate Destination Domain Enabled |
Select Yes to allow the administrator to specify the alternate domain to be used for this peer. Enter the alternate domain name. For example, when using the Microsoft Live Communications Server (LCS), the Office Communicator endpoint on the peer side may be in a different domain than the LCS. When enabled, this option will insert the alternate FQDN or IP Address into the To header. In addition, the FQDN or IP Address may be used to find the appropriate Peer Profile on incoming calls. |
No |
|
Alternate Destination Domain FQDN or IP address |
Enter the Fully Qualified Domain Name or IP Address of the Alternate Destination Domain. |
Blank |
|
Enable Special Re-Invite Collision Handling |
Select Yes to enable special handling of re-invite collisions. Normally, when a re-invite collision is detected, both re-invite messages are rejected with a 491. With this option enabled, the incoming re-invite wins. |
No |
|
Private SIP Trunk |
If enabled, calls received on this trunk will be considered private or non-public. The purpose of this option is to allow you to set SIP trunks to private for a small SIP gateway. Enabling this option allows the system to handle CPN substitution properly at ISDN and SIP interfaces. When trunks are connecting small gateways or devices that have non-public numbers, this option should be set to Yes (Private/non-Public). When set to Yes, calls delivered to call control will be treated as non-public trunk. If the call is directed to an ISDN type interface, the calling party number from a public trunk may pass "as is" out to the network. The private/non-public number may undergo CPN substitution before being sent out to the network. |
No |
|
Route Call Using To Header |
Enable this option to route all incoming calls to the called individual user based on the information present in the To URI instead of the Request URI. In most cases, the information in these two lines is the same. When the two lines differ, this option is used to select which one should be used. |
No |
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Calling Line ID Options |
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Default CPN |
Enter the default CPN up to 26 alphanumeric characters (a-z, A-Z and 0-9). This alphanumeric string is used in outgoing calls to replace the calling party number and on incoming calls to replace the called party number when a match is not found in the URI/Number Translation Form or the Outgoing DID ranges on the SIP Peer Profile Form. Normally on outgoing calls (unless restricted or private) the Default CPN would appear in the From header if a match is not found elsewhere. If the "Use P-Asserted-Identity Header" option is set, the P-Asserted-Identity will be included and it will contain the Default CPN if a match is not found elsewhere. A "Privacy: id" header will also be included if the CPN is restricted or the number is marked private. If the "Use P-Preferred-Identity Header" option is set and the "Use P-Asserted-Identity Header" is not set, the P-Preferred-Identity will be included and will contain the Default CPN. For incoming calls, the ringing or answering party's number is replaced with the default CPN when a match is not found in the URI/Number Translation Form or the Outgoing DID ranges. This number will only be included in the P-Asserted-Identity header if the option is enabled. To substitute specific numbers to DIDs, you may add numbers into the DID ranges for CPN Substitution form and then select the appropriate numbers in the Outgoing DID Ranges panel located at the bottom of this form (SIP Peer Profile).
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Blank |
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CPN Restriction |
Select to force anonymous@anonymous.invalid in the From header on outgoing calls and to prevent CPN substitution on incoming calls. If the "Use P-Asserted-Identity Header" option is set the P-Asserted-Identity will be included and it will contain the calling party number or some substitution. A "Privacy: id" header will also be included in this case to indicate that the identity should be kept private. |
False |
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Public Calling Party Number Passthrough |
If an incoming call is received by the 3300 ICP through an ISDN trunk or a Public SIP trunk you can allow the public number to be passed through the 3300 ICP when it leaves via a SIP trunk. Enable this option to allow the public CPN to be passed through the 3300 ICP and not substituted with the default CPN (normal behavior).
|
No |
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Use Diverting Party Number as Calling Party Number |
Enable this option to use the diverting/forwarding party as the CPN on the outbound SIP call instead of the original calling party number. The party at the final call destination sees the call as being from the diverting party because the original calling party information is not provided. |
No |
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Authentication Options |
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User Name |
Enter the authentication user name (up to 48 characters long) used to authenticate incoming/outgoing calls. The user name is also used to authenticate the registration, if applicable. |
Blank |
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Password |
Enter the authentication password associated with the Authentication User Name (hidden). The password is also used to authenticate the registration, if applicable. |
Blank |
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Confirm Password |
Re-enter the authentication password to confirm it. |
Blank |
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Authentication Option for Incoming Calls |
Select the type of authentication challenges for incoming calls:
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No Authentication |
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SDP Options |
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Allow Peer to Use Multiple Active M-Lines |
Enable this option for the 3300 ICP to allow the peer to negotiate more than one active m-line (media description line). For peers capable of negotiating multiple m-lines, such as audio, image, video, and applications, the recommended setting is enable or Yes. |
Disabled |
|
Allow Using UPDATE For early Media Renegotiation |
Select Yes to allow the media source to be switched prior to the call being answered.
|
No |
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Avoid Signaling Hold to the Peer |
Select Yes to prevent the 3300 ICP from sending Hold Indications to the peer. When this option is enabled, the 3300 renegotiates the SDP using "sendrecv" and not "sendonly" attributes. |
No |
|
Enable Mitel Proprietary SDP |
Select No to disable sending Mitel Proprietary lines within the SDP. Mitel proprietary SDP information is only of value when SDP from one 3300 ICP is delivered to another 3300 ICP. It is usually acceptable to select No when inter-working with 3rd party equipment. The other benefit of selecting No is that it reduces the SDP size. |
Yes |
|
Force sending SDP in initial Invite message |
Select Yes to always insert SDP into the initial Invite message. This option allows inter-working with SIP peers that require SDP in the Initial Invite. In many situations, the 3300 ICP already includes SDP in the initial invite. The forced SDP may contain a default inactive SDP if the SDP is not available at call setup time. |
No |
|
Force sending SDP in Initial Invite - Early Answer |
Enable this option if the Initial Outgoing Invite must contain a "sendrecv" Session Description Protocol (SDP) message that identifies the IP address/port of the calling device. This option takes precedence over the “Force sending SDP in initial Invite message” option (which may send an “inactive” SDP or an IP 0.0.0.0 SDP). The 3300 provides the connection information on the outgoing call by receiving a Fake Answer prior to sending the Initial Invite. |
No |
|
Limit to one Offer/Answer per INVITE |
Select Yes to ignore the SDP offer in 200 OK responses to the initial Invite message when media has already been established. Enable this option only if your service provider expects you to discard the SDP. Disable this option (select No) if your service provider expects a response. |
No |
|
NAT Keepalive |
Select to send UDP packets every 30 seconds to the Peer’s Audio IP Address and port to keep a pinhole open on a NAT firewall. The packets are sent whether the connection is one-way or two-way. |
False |
|
Prevent the Use of IP Address 0.0.0.0 in SDP Messages |
If enabled, a SENDONLY or INACTIVE SDP that is sent by the 3300 ICP will not contain the IP Address 0.0.0.0 as its connection address. If available, it will use the endpoint's IP address; otherwise, it will send the 3300's IP address + port 9000. Enable this option if there are specific 0.0.0.0 issues detected with the SDP. |
No |
|
Renegotiate SDP To Enforce Symmetric Codec |
Enable this option to force the use of the same codec--for example, G.729--in both incoming and outgoing directions. When a SDP negotiation is answered with a list of codecs it can be unclear which codec will be used in either direction. This option will send a Re-invite with a single codec if the 3300 ICP detects this situation to ensure there is no misunderstanding. |
Disabled |
|
Repeat SDP Answer If Duplicate Offer Is Received |
Select Yes to treat identical SDP offers received in the same session as a session refresh instead of responding with a new answer that repeats the previous SDP. By default, all SDP offers (duplicate or not) are treated as a new offer, resulting in audio renegotiation and the restart of RTP streaming. This option should be enabled if changing/restarting RTP streaming causes audio issues when the remote peer simply refreshes the media connection. |
Disabled |
|
RTP Packetization Rate Override |
Enable this option if the Service Provider to which this peer is connected requires a packet rate other than the standard 20 ms rate in both the transmit and receive media streams. The following Mitel devices and applications will support variable packetization rates:
|
Disabled |
|
RTP Packetization Rate |
If the "RTP Packetization Rate Override" option has been set to Yes, the SIP Module will force all calls involving this trunk to use the packetization rate specified in the "RTP Packetization Rate" option. The calls will be forced to use the specified rate in both the transmit and receive streams. Rejected calls or audio problems resulting from an unsuccessful packet rate negotiation will generate Media Negotiation maintenance logs. For more information, see Maintaining the SIP Interface. |
20 ms |
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Special handling of Offers in 2XX responses (INVITE) |
Set to Yes to have the 3300 ICP treat 2XX "sendonly" offers as "sendrecv" offers, causing a data stream to be opened to the IP address and port specified in the sendonly offer. For this option to work, the incoming messages must have a valid IP address and port number (not zeroes). This option addresses an interoperability problem with some SIP gateways, whereby unending message negotiation causes an infinite loop.
|
No |
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Suppress Use of SDP Inactive Media Streams |
Select Yes to allow the 3300 ICP to minimize the use of INACTIVE messaging if it is not fully supported by the service provider. While media connections are transitioning (hold/transfer/conference, etc.), or in some cases at call setup, the media may be temporarily unavailable causing the 3300 ICP to send an inactive SDP. If this option is set to Yes the 3300 ICP will instead attempt to use the IP 0.0.0.0 or mark streams as sendonly/recvonly to avoid the use of inactive which is not supported by some SIP Peers. |
No |
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Signaling and Header Manipulation Options |
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Session Timer |
Set the SIP session timeout value (in seconds). If the peer does not respond within the allocated time, the session (call) will be torn down. The default is 90. The range is from 90-9999. Setting the Session Timer to 0 disables session timeout. It is recommended that this be set to a non-zero value unless there is a specific reason to disable it. The benefit of this option is that it can help clear calls that get stuck due to signaling errors. If the peer responds with a 422 “Session Interval too Small,” you may want to increase the session timer to the value indicated by the peer to minimize delays in call setup. |
90 |
|
Allow Display Update |
Select Yes to allow the system to signal display updates containing connected party name and number information during transfers, conferences, and call forwarding scenarios. The information is sent in Update messages (as defined in RFC 4916) or in the P-Asserted-Identity header (if configured).
|
No |
|
Build Contact Using Request URI Address |
Enable this option to construct the contact address in 180 and 200 messages using the Request URI Address received in the initial invite. This option should only be enabled in those situations where the Contact needs to be based on the address received in the Request URI. |
No |
|
Disable Reliable Provisional Responses |
Select Yes to disable the use of reliable provisional responses (PRACK) on outgoing and incoming calls, unless the Required Header is received on incoming calls. Most Peers now support PRACK and this can be useful in interoperability scenarios with the PSTN (see RFC 3262). If the SIP Peer also supports PRACK, it is recommended that this option be set to No. |
Yes |
|
Enable Sending '+' for E.164 numbers |
Select Yes to enable adding '+' to the Called and Calling Party Numbers generated by the 3300 ICP. |
No |
|
Ignore Incoming Loose Routing Indication |
Select Yes for the 3300 ICP to ignore the loose routing indicator and use strict routing instead. Enabling this option is not recommended unless required by the SIP Peer. See RFC 3261 for more information on Loose Routing. |
No |
|
Use P-Asserted Identity Header |
Select Yes to enable sending the P-Asserted Identity header within SIP messages. This option is used to convey identity information both at call setup and during a call. When privacy is enabled, or CPN is restricted, this field will still contain calling/called party information. The 3300 ICP relies on the use of the “Privacy: id” header to inform other SIP Peers that this information is to be kept private. If the peer is not trusted, select No for this option. |
No |
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Use P-Preferred Identity Header |
If you enable this option, the system uses the Default CPN data to build the P-Preferred-Identity header. No display name is shown in this header. A P-Preferred-Identity will not be included in messaging if the “Use P-Asserted-Identity” option is also enabled. One purpose of this header is to provide a company or link/peer number. For example, the From header may include the DID of the person making the call but the P-Perferred-Identity header may contain the main company DID. |
No |
|
Use Restricted Character Set for Authentication |
Select Yes to use the restricted character set "0123456789abcdef" for creating nonce and cnonce strings used in authentication. This setting is used for SIP Servers with limited ascii character support and reduced authentication security. |
No |
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Use To Address in From Header on Outgoing Calls |
When this option is enabled (set to Yes) the Address used in the SIP From header line will no longer be the physical 3300 IP address or FQDN. Instead the address will be replaced by the address to which the outgoing call is sent. Some providers require this for authentication purposes as it makes it look like the 3300 ICP is in the same domain as the SIP Server or SBC (Session Border Controller). |
No
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Index |
Enter an index number between 1 and 100. Index numbers refer to substitutions programmed in the DID Ranges for CPN Substitution form. If a DID number sent over this link matches the DID range programmed for this index number, then the system makes the substitution. |
Blank |