SIP Private Networking is the network protocol of choice when no QSIG legacy PBXs are in the customer network. This eliminates the need to convert between SIP and SIP-Q protocol for a station-to-station call between two Lotus Sametime Unified Telephony systems. SIP Private Networking is sometimes also referred to as Enterprise SIP Trunking or Enterprise SIP Peering.
When the originating interface (i.e. SIP subscriber or SIP trunking endpoint) is SIP, the SIP Private Networking interface shall be used when the call is routed to another Telephony Control Server or the SIP Trunking interface when the call is routed to a gateway, configured for interworking SIP to a non-QSIG protocol (e.g. RG8700).
When the originating interface is a SIP-Q Private Networking endpoint, then SIP-Q Private Networking shall be used when the call is routed to another Telephony Control Server or gateway, configured for interworking SIP-Q to QSIG (e.g. RG8700, or HiPath 4000).
The basic rule is to provision trunking routes through the private network with the same protocol as used for the incoming call request at the originating Telephony Control Server system. This way, one or no conversion between SIP and SIP-Q occurs on the end-to-end path of any one call.
Figure 1. Routing principles for SIP/SIP-Q Routing
In order to achieve SIP-Q equivalency, the following SIP-Q functions/features must be available on the SIP interface to create the SIP Private Networking interface.
Transport of CDR information (e.g. Global ID, Thread ID)
Call Transfer Information to support proper CDR records in originating, transit and terminating Telephony Control Server systems.
Call Diversion Information to support privacy in diversion headers in forwarding, transit and forwarded-to Telephony Control Server systems.
Parent topic: Networking
- Identification Information to support proper CDR records and proper displays.