Using the Session Initiation Protocol (SIP) at the application layer, SIP trunks bundle multiple communication channels into a single logical connection that can support hundreds — or even thousands — of concurrent sessions without degrading quality. Media is carried using the Real-Time Transport Protocol (RTP), ensuring voice and video packets traverse the network with predictable latency, jitter, and loss characteristics.
With SIP trunking, you can:
- Connect directly to cloud communication platforms like Microsoft Teams, Zoom, Cisco Webex, and leading Contact Center as a Service (CCaaS) solutions
- Extend the life of existing PBX investments while enabling global reach
- Reduce operational costs by eliminating legacy TDM trunks and long-distance fees
What is SIP Trunking?
SIP trunk definition: A virtual, software-based communication channel that connects an IP-enabled PBX to other IP-based endpoints or the public switched telephone network (PSTN) via a data network. This transport path eliminates the need for copper wires and extends the range of locations that can be connected.
The SIP protocol manages each communication session independently, while the Real-time Transport Protocol (RTP) carries the stream of media packets over IP. SIP enables IP-based calls that flow across the same data backbone as your other digital workloads. And because it handles voice and video together, calls can utilize a combination of channels without the need to change platforms.
How Does SIP Trunking Work?
There are four basic steps that define SIP trunk:
- Call setup: The IP-enabled PBX (via softphone, IP phone, or mobile device) sends an INVITE request to a Session Border Controller (SBC). This request specifies the caller and codec details for the call.
- Session negotiation: The SBC then exchanges SIP messages with the carrier’s core to allocate a channel and relay authentication.
- Media flow: Data packets containing the voice, video or text message travel via RTP across the appropriate available data network.
- Teardown: When the call is complete, a BYE request is sent, releasing the channel and opening the bandwidth for other calls.
What is SIP Trunking Used For?
SIP trunking replaces your existing telephony infrastructure, then combines it with your video conferencing and text messaging needs into one consolidated, flexible and scalable solution that’s ready to grow with your enterprise. An example of some specific uses includes:
- Cloud connectivity: By connecting directly to your existing cloud contact center solutions, trunking allows enterprises to have a single carrier globally.
- Legacy PBX modernization: Existing legacy calling hardware like Avaya, Cisco, or NEC PBX systems can have their useful lifetimes extended by connecting them to a SIP trunk.
- Global number management: SIP trunking allows enterprises to easily provision direct inward dialing (DID) numbers without having any in-country facilities, supporting seamless omnichannel customer experiences.
- Carrier consolidation: One SIP trunk provider replaces every local telco you’re currently using, simplifying billing, invoicing, plan administration and regulatory compliance.
Benefits of SIP Trunking
Reliability
Reduced Expenses
Scalability
How to Switch to SIP Trunking
A broad set of steps for switching from PSTN to SIP trunking should include:
- Infrastructure audit: A full cataloging of your existing PBX, including models, firmware versions, codecs in use and available WAN bandwidth provides a starting point and lets you identify hardware that needs upgrading and identify bandwidth requirements.
- Select provider: Time spent researching provider options, comparing SLAs, global DID availability and options for redundancy will be time well spent. This background helps ensure you find the provider that can provide the coverage and service quality you need.
- Pilot deployment: A test deployment will allow you to tune quality of service, alert you to any VLAN adjustments you’ll need to make and give you a benchmark for packet capture.
- Stage migration: Porting numbers in blocks while maintaining a dual PSTN/SIP set-up for failover provides business continuity while you transition fully to your new SIP trunking infrastructure.
- Monitor and optimize: Tracking appropriate SBC analytics and making adjustments early will provide a foundation that will help sustain call quality and security post-deployment.
How is SIP Trunking Priced?
Pricing for SIP trunking will vary depending on many factors. Some of these factors will include:
- Size of enterprise
- Geography (where HQ is located and how many global locations you’re connecting)
- Call volume
- Plan type
- Metered (pay as you go)
- Unmetered (per channel)
- Hybrid infrastructure
- Upgrades necessary (is your existing PBX IP-enabled, for example)
Make the Move to the Cloud with GTT
GTT operates one of the world’s largest Tier 1 IP backbones, connecting more than 600 locations and delivering low-latency SIP trunks that cover five continents. Some key differentiating features of GTT SIP trunk offering include:
- True global footprint: Local DIDs in 3,700+ cities spanning 65+ countries
- Secure trunking: End-to-end TLS encryption and managed SBCs
- Cloud-enabled: Pre-established secure connections to Teams, WebEx Calling, Zoom Phone and Anywhere365
SIP trunking is a strategic option for consolidating voice calling, video conferencing and text messaging into one unified, secure offering. Trunking cuts overhead, simplifies number administration and enables more robust business continuity. When you’re ready to modernize your business’s voice infrastructure, connect with a GTT voice expert to start mapping your migration.
FAQs ABOUT SIP Trunking
What is the difference between VoIP and SIP trunking?
VoIP is a general term for a collection of technologies that allow voice communication over data networks. SIP trunking is one implementation that uses the SIP protocol to bridge on-premises or cloud PBXs to the PSTN.
What are the benefits of trunking ports?
Trunk ports aggregate traffic from across multiple VLANs, optimizing bandwidth allocation and reducing latency for voice and video traversing the network as data packets.
How much does SIP trunking cost?
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