{"id":51,"date":"2026-04-07T16:23:15","date_gmt":"2026-04-07T16:23:15","guid":{"rendered":"http:\/\/localhost:8080\/knowledge-base\/outbound-registration-connections\/"},"modified":"2026-04-18T13:26:14","modified_gmt":"2026-04-18T13:26:14","slug":"outbound-registration-connections","status":"publish","type":"post","link":"https:\/\/www.siperb.com\/kb\/outbound-registration-connections\/","title":{"rendered":"Outbound Registration Connections"},"content":{"rendered":"\n

Outbound Registration involves entering SIP details from your PBX or ISP into SIPERB systems to maintain a registration loop through REGISTER messages. For connectivity, server details like address and port are crucial. SIPERB acts as a proxy, with calls initiated from the client side that include all details for a media session. Your firewall must allow inbound registration messages from the SIPERB SBC.<\/p>\n\n\n\n

Dial Pattern :<\/strong> The dial pattern is the regular expression used to match against 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 .<\/code> (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}$<\/code> and the connection to Office B with a dial pattern of ^3d{2}$<\/code>. You could even have a third connection that is connected to an ISP with a catch-all dial pattern (.<\/code>), but then you should make sure that it has a greater Weight<\/strong> than the other connections.<\/p>\n\n\n\n

Weight :<\/strong> 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 \u201ccatch-all\u201d Dial Pattern (.<\/code>) should have the highest value.<\/p>\n\n\n\n

Transcoding<\/h2>\n\n\n\n

Whether transcoding is required depends on the transport<\/strong> you have selected for this connection. Call Transcoding is available on all plans, including free.<\/p>\n\n\n\n

UDP or TCP \u2014 transcoding is automatic<\/h4>\n\n\n\n

When your connection uses UDP or TCP<\/strong>, 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 \u2014 Siperb handles the codec conversion transparently, with no additional configuration required on your part.<\/p>\n\n\n\n

TLS \u2014 transcoding is optional<\/h4>\n\n\n\n

When your connection uses TLS<\/strong>, SIP signalling is encrypted between Siperb and your PBX. If your PBX also supports DTLS-SRTP<\/strong> for media (i.e., it is WebRTC-capable), you can disable transcoding. This creates a fully encrypted, end-to-end media path \u2014 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<\/strong> to TLS and leave Transcoding disabled<\/strong>.<\/p>\n\n\n\n

If your PBX uses TLS for signalling but expects plain RTP for media, leave transcoding enabled \u2014 the SIP signalling will be encrypted, but Siperb will still bridge the media.<\/p>\n\n\n\n

For a deeper explanation of why transcoding is needed and when you can avoid it, see Call Media Transcoding \u2192<\/a> and WebRTC End-to-End Encryption \u2192<\/a><\/p>\n\n\n\n

When transcoding is active \u2014 firewall details<\/h4>\n\n\n\n

IP Address :<\/strong> 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.<\/p>\n\n\n\n

Port:<\/strong> 10000 \u2013 65000. We suggest opening the entire UDP range greater than 10000 to and from our Transcoding Server(s).<\/p>\n\n\n\n

Protocol:<\/strong> (UDP) Media will always be in UDP protocol but the underlying media encoding will be set according to the media transcoding options.<\/p>\n\n\n\n

SBC Server<\/strong><\/h2>\n\n\n\n

The Session Border Controller (SBC) is the public facing interface of the Siperb Network. It\u2019s the first of 3 Proxy Servers that allows calls to flow form 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.<\/p>\n\n\n\n

Register Binding SBC :<\/strong> You can choose the SBC server by selecting one from the dropdown above. This list may expand over time.<\/p>\n\n\n\n

Binging Type:<\/strong> Select how you want to bind the SBC.<\/p>\n\n\n\n