Create SIP Trunk
Step 1. In the left sidebar menu, navigate to Voice and select Inbound Trunk. To create a new SIP Trunk, click Create New and then select SIP Trunk. (Fig. 1)
Fig. 1. Adding a new SIP Trunk
Step 2. You will be prompted to enter your SIP Trunk details (Fig. 4).
Fig. 4. SIP Trunk configuration details window
Basic configuration options:
Friendly name - a desired trunk name.
Username - the user part of R-URI in the INVITE request. The following placeholders can be used in the username field:
{DID} - a pattern which will be replaced by the called DID number (DNIS) in E164 format.
{CALL_CPC} - used to identify the type of the calling party making the call.
Note
CPC usage can be found here
Host - your server’s public IP address or domain.
Port - SIP port on your server. In order to use DNS SRV failover mechanism leave port empty. In case of empty value DIDWW system will try to resolve DNS SRV records, then A record will be tried (with default SIP signaling port 5060)
Preferred server - the preferred DIDWW POP server, with the options being US: LA, MIA, NY, Germany: FRA, SG (Singapore), HK (Hong Kong) or Auto.
Allows to select specific POP through which calls should be routed towards set end destination.
Selecting Auto preferred option is highly recommended. With Auto option the system will select the best POP to route calls towards destination according to the POP from which call has been received. Auto preferred server selection provides the best performance.
The following figure (Fig. 2) illustrates incoming call being received from PSTN network to DIDWW HK SBC and being routed to DIDWW FRA SBC when FRA Preferred server option is selected. The routing according to the configuration causes additional hop being present until the call reaches customers equipment.
Fig. 2 Preferred Server: FRA
With the following figure (Fig. 3) which illustrates routing using Preferred server option Auto is routing the call from the same HK DIDWW SBC as the call was received from PSTN network.
Fig. 3 Preferred Server: Auto
Warning
When Preferred server option Auto is selected, the call might arrive from any of the available SBCs. It is recommended to have all our signaling and RTP IPs configured within customer’s equipment.
Note
When using Preferred server option Auto, please enable all available POPs in Settings as not enabling all POPs may cause routing over additional hops as shown in Fig. 4.
Transport Protocol - the underlying transport layer protocol (UDP or TCP) that will be responsible for SIP requests and responses.
Network Protocol - determines which IP protocol version will be used for communication between the SIP trunk and the endpoint. The available options are:
IPv4 only: Use only IPv4 protocol for the SIP connection. Select this if your network and endpoints support only IPv4.
IPv6 only: Use only IPv6 protocol. Choose this option if your network and SIP infrastructure are configured for IPv6.
Any: The system will use either IPv4 or IPv6, depending on what is available and configured in your network.
Prefer IPv4 over IPv6: The system will prefer IPv4 if both IPv4 and IPv6 are available. It will fall back to IPv6 only if IPv4 is not accessible.
Prefer IPv6 over IPv4: The system will prefer IPv6 if both IPv4 and IPv6 are available. It will fall back to IPv4 only if IPv6 is not accessible.
Capacity Limit - the maximum number of simultaneous calls per trunk.
Resolve rURI - if checked, host part or R-URI will be replaced by the resolved IP address.
Resolve DNS SRV record - if checked, the system will attempt to perform a DNS SRV lookup for the host part of the R-URI to override the port settings.
CLI Format - the following format options are available:
raw: DIDWW will pass the incoming caller ID unchanged (default).
e164: DIDWW will attempt to convert the CLI to E.164 format (Country Code + Area Code + Number)
local: DIDWW will attempt to convert the CLI to local format (Area Code + Number)
Note
CLI format conversion may not work correctly for phone calls originating from outside the country of the specific DID
CLI Prefix - a field that allows prepending an additional prefix to the incoming CLI (maximum of 7 characters)
Assign trunk to all DIDs - if checked, then all DIDs in your account will be assigned to this inbound trunk.
Step 3. Click “Create” to complete the trunk configuration.
SIP Technical information is available here.
In addition to the basic configuration options, there are advanced options which may be left unchanged, or recommended values can be set by pressing “Use recommended values” button. These options include:
Trunk Group Configuration
Authentication
Fig. 5. Authentication configuration window
- Optional authentication for this SIP trunk, with parameters:
Auth enabled - if checked, then authorization for the SIP server is enabled
Auth user - optional authorization user for the SIP server
Auth password - optional authorization password for the SIP server
Auth “From” User - Specifies user in a “from:” field instead of the Caller ID
Auth “From” Domain - Specifies the default “From” domain in SIP messages
Media & DTMF
Fig. 6. Media & DTMF configuration window
- Optional signalling configuration with parameters:
Available codecs/Selected codecs - codecs that will be sent with an SDP offer may be selected from the list of supported codecs. Codecs can be added and removed by selecting them from the list and clicking the middle arrow buttons.
Customer To DIDWW DTMF format - the method of receiving DTMF signals from the CPE (options are RFC 2833, SIP INFO application/dtmf-relay OR application/dtmf, or RFC 2833 OR SIP INFO)
DIDWW to Customer DTMF format - the method of sending DTMF signals to the CPE CPE (options are disable sending, RFC 2833, SIP INFO application/dtmf-relay, or SIP INFO application/dtmf)
RTP Timeout - timeout (seconds) for disconnecting the call if RTP packets are not received
RTP Ping - if checked, an RTP ping should be used for connecting the call
Force Symmetric RTP - if checked, the trunk will operate in the Symmetric RTP/COMEDIA mode
Symmetric RTP Ignore RTCP - if checked, then only RTP packets will be considered while operating in the Symmetric RTP/COMEDIA mode, and switching of the RTP session based on RTCP packets will be avoided
Media Encryption Mode - Our systems supports TLS as a secure transport for SIP signaling and SRTP as a media encryption mechanism. All three SRTP key negotiation mechanisms are supported - SDES, DTLS and ZRTP.
Warning
The SRTP media encryption is disabled by default. Please contact our sales team sales@didww.com to enable SRTP support for your account.
CNAM IN
Fig. 7. CNAM IN configuration window
Inbound CNAM lookup - enabling this feature will display the caller’s name associated with US phone number
When the feature is enabled, the system will perform CNAM Lookup in the remote database when Caller ID is from US
CNAM Lookup is being billed if the CNAM Lookup is successful
The CNAM name is included in FROM header, Display Name
Warning
When a Trunk Group is used and any of trunks has CNAM IN feature enabled, CNAM Lookup will be performed and a successful Lookup will be billed. CNAM Lookup value will be displayed for trunks that has CNAM IN feature enabled.
Example of FROM header when CNAM IN feature is not enabled:
From: "12899230448" <sip:12899230448@46.19.213.14>;tag=24-6917EC3D-6453984E000C1C1E-5EA5B700
Example of FROM header when CNAM IN feature is enabled and CNAM Lookup is successful:
From: "KRIS TOTAL" <sip:15203981500@46.19.213.14>;tag=24-7EB42886-645381060002B85A-5EB5C700
Example of FROM header when CNAM IN feature is enabled, but CNAM Lookup has failed:
From: "Unavailable" <sip:15203981500@46.19.213.14>;tag=24-7EB42886-645381060002B85A-5EB5C700
CNAM IN value is also displayed in Source Name field in Call Logs, Inbound section.
Advanced Signalling Settings
Fig. 8. Avanced Signalling Settings configuration window
- Optional SIP configuration parameters as follows:
SST enabled - (RFC 4028) if checked, then the customization of SIP Session Timers is enabled in order to ensure that a session remains alive
SST Accept 501 - if checked, then the call will not be dropped after receiving SIP 501 response for non-critical messages
SST MIN Timer - (RFC 4028) minimum SIP session timer value (default 600 seconds)
SST MAX Timer - (RFC 4028) maximum SIP session timer value (default 900 seconds)
SST Session expires - (RFC 4028) Optional ‘Session-Expires’ header values (should be in range SST MIN Timer to SST MAX Timer)
SST Refresh method - (RFC 4028) SIP method that is used for session update (options are Invite, Update and Update Fallback Invite)
SIP Timer-B - (RFC 3261, Section 17.1.1.2) INVITE transaction timeout (default 8000 ms)
DNS SRV failover timer - Invite transaction timeout for each of the gateways with DNS SRV rerouting (default 2000 ms)
Max Transfers - maximum REFER Request count
Max 30x Redirects - maximum 301/302 SIP Redirect count
STIR/SHAKEN
Optional caller-id attestation headers based on the new STIR/SHAKEN policy. See Voice IN STIR/SHAKEN page for details.
CPC usage
The CPC values are assigned to each source type:
01 – cpc=ordinary, 02 – cpc=cellular, 03 – cpc=payphone, 04 – cpc=unknown.
Note
CPC values are supported for Toll-Free numbers only
E.g. SIP URI: {DID}{CALL_CPC}@my.domain.com
Incoming call to DID 18553864279, coming from:
Landline source: 123456789001@my.domain.com
Mobile source: 123456789002@my.domain.com
Payphone source: 123456789003@my.domain.com
Unknown source: 123456789004@my.domain.com