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SIP Trunking
A Session Initiation Protocol (SIP) to PSTN gateway may have trunks that are connected to different carriers. It is entirely reasonable for a SIP proxy to choose — based on factors not enumerated in this document — which carrier a call is sent to when it proxies a session setup request to the gateway. Since multiple carriers can transport a call to a particular phone number, the phone number itself is not sufficient to identify the carrier at the gateway. An additional piece of information in the form of a trunk group can be used to further pare down the choices at the gateway. As used in this document, trunks are necessarily tied to gateways, and a proxy that uses trunk groups during routing of the request to a particular gateway knows and controls which gateway the call will be routed to, and knows what trunking resources are present on that gateway.
In an architecture where calls can be terminated on multiple gateways it is wise to consider routing the call to a destination based on some significant criteria such as cost, quality or proximity. Where a proxy has the ability to evaluate a call based on one or more of these criteria, as well as knowledge of the TDM trunk resources available, the proxy can "tag" the call using the tgrp and trunk-context values in the SIP Contact field of the INVITE. It is important to note that the tgrp and trunk-context values can only be used with a TEL URI, not with a SIP URI.
Unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these traditional fixed Public Switched Telephony Network (PSTN) lines with PSTN connectivity via a SIP trunking service.
To be added...
- PSTN-to-PSTN calls traversing SIP networks
See Also:
SIP | RFC4904Created by: SCR,Last modification on Mon 09 of Aug, 2010 [01:59 UTC]
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