When connecting to an ITSP (Internet telephone service provider), you may be required to use a SIP registration trunk due to the fact that the ITSP does not need to know your public IP address. They will learn it through the registrations process. The SIP registrations trunk is a popular type of SIP trunk and is easy to set up once the ITSP account has been established. It requires the system to register to the provider, just like an IP phone registers with the system. Like an extension that has been registered with the system, SIP registration trunks require a user name or account (usually the DID), a password, and the IP address or domain name of the SIP or proxy server. The advantage of the SIP registrations trunk is that the IP address is dynamically bound to the SIP registration, which allows the registration to be used from any IP address. Although service providers often assume that a trunk registration involves an IP phone or an ATA rather than a trunk, many phones can sit behind the snom ONE telephone system and share the resources of the trunk (see image). However, most service providers limit the number of calls that can be made over a SIP trunk and use a charging method that simulates the charges accumulated by a physical legacy TDM trunk (i.e., the number of call paths).
Although SIP trunks are virtual trunks and can theoretically have an infinite number of calls, SIP trunks are limited by the amount of bandwidth that is available to handle the calls.
A SIP registration trunk can also be used to connect two or more snom ONE systems together, as in a branch/head office configuration (see Connecting Branch Offices Together).
The gateway trunk is typically used to talk to a PSTN or cellular gateway, which could authenticate the call leg on the IP address in the system. However, some ITSPs that do not have a session boarder controller and require a public IP address use gateway trunks. A gateway is defined as a device that converts media from one network or protocol to another. This type of trunk is used for a media gateway.
SIP gateways are used to transport public switched telephone network (PSTN) terminated signaling across an Internet protocol (IP) network. The gateway sits between the PSTN and the IP network (see image). Unlike the SIP registrations trunk, the gateway model does not register—It just sends the traffic to the destination. In this model, the system uses the caller-ID of the system to indicate the extension that initiated the outgoing call (if that extension did not block the caller-ID). This model is typically used with PSTN gateway hardware located on customer premises, but it can also be used to link two snom ONE systems together as long as they are routable to each other, i.e., both are on public IP addresses or on the same private network.
The outbound proxy trunk can be used to communicate with any other type of SIP proxy or IP telephone system or to join two IP telephone system deployments together. The outbound proxy trunk is a direct connection to the network and is similar to the gateway model. The difference is in the way anonymous calls are made and in how the proxy represents its own domain. As the name suggests, the proxy model assumes that you are talking to a SIP proxy or session border controller, while the gateway model assumes you are talking to a SIP user agent. However, the two models are quite similar. As a general rule, use the gateway trunk first, and if you have an issue, try changing the trunk type.
WebRTC trunks are using the web browser for generating the voice packets. For more information, see WebRTC Trunk.