SIP Trunking uses VoIP to connect a PBX between the Internet and the Public Switched Telephone Network (PSTN), replacing a traditional “phone trunk” such as a Primary Rate Interface (PRI) or analog line. This solution requires an on-premise PBX and a gateway to connect your Internet telephony service provider to a PBX.
Trunking to a Hosted PBX is typically done using SIP.
SIP Trunking’s primary functions include:
- Locating the User
- Selecting the end system for a session
- Learning user availability
- Determining the capability of the end-user system
- Establishing a session(call)
- Managing the call session, including termination, transfers and more
Pros and Cons
Leverages your IP Network, turning voice into an application on the network.
Potential for improved cost efficiency and cost savings.
Additional call appearances can be added quickly without having to wait for more circuits to be installed.
Call appearances can be routed to other sites quickly so you have flexibility with where phone service is being provided.
Effective bandwidth analysis to protect QoS is especially important, due to multimedia transmissions.
Can require higher investment costs, due to needing to acquire new equipment and retire old equipment.
The newness of this technology can make finding talent and troubleshooting help more challenging.