SIP TRUNKING: WHAT IS IT?
A SIP (Session Initiation Protocol) Trunk is a telecommunication service that connects a company’s premise-based PBX, conferencing platform, IVR or other IP-based telephony platform to the PSTN via the Internet or other network transport, using SIP and RTP.
SIP is the signaling layer for call set-up and tear-down. RTP (Real-time Transport Protocol) is the voice channel. The industry generically calls this service SIP Trunking, but is also referred to as Voice Trunking or Voice Gateway Services.
General Benefits of a SIP Trunk
- Cost Savings: Businesses can generally save anywhere from 50% to 90% of their PRI bill with reduced costs over traditional TDM Voice Trunks.
- Flexibility: Easily add new IP-based features that are not available over traditional TDM equipment and do so without the need for additional equipment, such as add Call Recording. More easily add new phone numbers and features.
- Scalability: SIP Trunking is not bound by physical channel limits like traditional TDM circuits. SIP trunks can more easily scale up on demand as needs arise.
- Easier to Manage: No need to always contact your provider to make changes to your account. Often these changes take weeks to complete if not longer. Self-service options are now available vs. traditional carrier-only model.
- Improved Voice Quality: Take advantage of high def codecs that are only available over IP-based connections.