Outbound Trunk Connections

Outbound Trunks function similarly to Outbound Registration but do not send a REGISTER message, meaning there’s no registration status available. They are commonly used when connecting to endpoints with a static or fixed IP address since the primary purpose of registration is to retrieve an IP address.

Dial Pattern : The dial pattern is the regular expression used to match again the number you are dialling out, allowing this connection to be selected when a particular number is dialed. The broadest possible match is simply a . (a full stop). You can have multiple connections and with each connection they can match on a different regular expression. For example; if you have two connections, one to Office A where your extension numbers all in the 200 to 299 range, and another connection to Office B where your extensions are all 300 to 399, then you would set the connection to Office A with a Dial Pattern of ^2d{2}$ and the connection to Office B with a dial pattern of ^3d{2}$. You could even have a third connection that is connected to an ISP with a catch-all dial pattern (.), but then you should make sure that it has a greater Weight than the other connections.

Weight : The Weight is used if you have more than one connection. The Weight value is a numeric value to determine the order of connections when performing the dialed number lookup. The lookup is performed in a loop starting with the lowest Weight, and moving higher, and when a match is found, the test ends. Typically a “catch-all” Dial Pattern (.) should have the highest value.

Transcoding

Whether transcoding is required depends on the transport you have selected for this connection. Call Transcoding is available on all plans, including free.

UDP or TCP — transcoding is automatic

When your connection uses UDP or TCP, your PBX communicates with Siperb over standard unencrypted SIP. Because the Siperb client uses encrypted WebRTC media (DTLS-SRTP with Opus) and your PBX expects plain RTP, Siperb must bridge the two. Transcoding is enabled automatically for UDP and TCP connections and cannot be disabled — Siperb handles the codec conversion transparently, with no additional configuration required on your part.

TLS — transcoding is optional

When your connection uses TLS, SIP signalling is encrypted between Siperb and your PBX. If your PBX also supports DTLS-SRTP for media (i.e., it is WebRTC-capable), you can disable transcoding. This creates a fully encrypted, end-to-end media path — audio travels as DTLS-SRTP from your softphone all the way to your PBX, with Siperb relaying the stream without decoding it. To use this mode, set Transport to TLS and leave Transcoding disabled.

If your PBX uses TLS for signalling but expects plain RTP for media, leave transcoding enabled — the SIP signalling will be encrypted, but Siperb will still bridge the media.

For a deeper explanation of why transcoding is needed and when you can avoid it, see Call Media Transcoding → and WebRTC End-to-End Encryption →

When transcoding is active — firewall details

IP Address : This is the live IP address of the selected Transcoding Server. We allow you to select an available transcoding server so that you can open this IP to media traffic inbound and outbound on your firewall.

Port: 10000 – 65000. We suggest opening the entire UDP range greater than 10000 to and from our Transcoding Server(s).

Protocol: (UDP) Media will always be in UDP protocol but the underlying media encoding will be set according to the media transcoding options.

SBC Server

The Session Border Controller (SBC) is the public facing interface of the Siperb Network. It’s the first of 3 Proxy Servers that allows calls to flow from your network through ours and on to your devices. We allow you to choose a different SBC for each of your own connections. Once a connection is setup on an SBC, it can be changed by selecting another SBC from the list. The choice of SBC is important for you to understand, as it determines the source or destination of the signalling traffic. This will be important when you configure you Asterisk (or other) PBX.

Host Address : You can choose the SBC server by selecting one from the dropdown above. This list may expand over time.

IP Address : This is the live IP address of the selected SBC. You may want to make note of this to allow SIP traffic in and out of your firewall.

Note: Never send a call to us as an IP address, always send your INVITE as a full domain INVITE <sip:123456789@eu-west-1-sbc-1.siperb.com> SIP/2.0.

Port: (5060) We will send SIP messages on port 5060, and you can send SIP messages to us on port 5060.

Protocol: (UDP) We will send SIP messages on an UDP protocol, and you can send SIP messages to us on an UDP protocol.

Registration

Registration is an important part of your connection. With the Outbound Trunk, there will be no REGISTER loop from our servers, and the auth details will only be presented once challenged by you (or your ISP).

Note: It is vital to use exact IP matching in the Allow Source option above as calls inbound are not challenged for auth.

End Point Server : The End Point Server is the Server to which you would like to send your call.

Username : Provide the auth username to present when challenged in Digest authentication with your server (or ISP).

Password : Provide the auth password to present when challenged in Digest authentication with your server (or ISP).

Allow Source IP/Subnet : The allow source IP or Subnet is a list of IP addresses (eg “192.168.1.0”) or Subnets (”192.168.0.1/24”), that restricts access to your account when a call comes in. Each inbound call on our SBC is inspected and matched against an account, once the match is found, the related connection is loaded and this property is matched against the IP represented as the source of the message. If there is a match the call is allowed, otherwise it’s rejected.

Note: this option can be set to 0.0.0.0/0 (all possible IP addresses), but this should only be used with Inbound Registration’s as they are challenged with Digest Authentication. It’s recommended to always make use of this option.

Realm : This is the realm that you have configured on your server (or ISP) as part of the endpoint auth configuration. Typically this is asterisk on Asterisk servers, but can also be set to * for the system to adopt the presented realm.