Interest in this protocol isn't fading. Metrigy pegs the global UCaaS market at $21.7 billion in 2024, a 6.5% jump year-over-year, with growth projected through 2029. Cloud communications keeps expanding, and SIP sits underneath most of it.
Yet SIP gets confused with VoIP constantly, and plenty of people assume it carries the actual audio. It doesn't. This guide walks through the real sequence: registration, call setup, media exchange, and termination.
Key Takeaways
- SIP is the signaling protocol that starts, manages, changes, and ends a call session.
- RTP carries the actual voice audio once SIP finishes negotiating the session.
- Five stages define the full call path: registration, INVITE, response and negotiation, audio exchange, then BYE.
- Network setup, codecs, device readiness, and provider infrastructure determine call quality.
What Is SIP Calling?
SIP calling means using Session Initiation Protocol to set up and manage voice, video, or messaging sessions over an IP network. Defined in RFC 3261, it's an application-layer signaling protocol built to create, modify, and terminate sessions between endpoints.
The problem it solves is practical: businesses need to connect callers and phone systems without running a dedicated physical line for every single call. SIP replaces that hardware dependency with signaling messages that travel over existing internet connections.
SIP vs. VoIP vs. SIP Trunking
These three terms get used interchangeably, but they're not the same thing.
| Term | What It Actually Is |
|---|---|
| VoIP | The broad method of transmitting voice over IP networks |
| SIP | A specific signaling protocol used to control those sessions |
| SIP Trunk | The virtual connection linking a PBX to a provider or the PSTN |
SIP is not a phone number, a physical cable, a complete phone system, or the audio stream itself. It's the traffic controller, not the traffic.
The Moving Parts
A typical SIP call involves several components working together:
- The caller's SIP endpoint (phone, softphone, or app)
- A SIP server or proxy that routes requests
- A registrar that tracks where endpoints can be reached
- The receiving endpoint
- The PBX or hosted phone system
- The separate media path that carries audio
Whether a business runs a hosted cloud PBX, an on-premises PBX with SIP trunking, or softphone calling through a mobile app, the signaling sequence stays largely the same. Only the equipment underneath changes.
How Does a SIP Call Work?
SIP signaling establishes the conversation. A separate media protocol, almost always RTP, transports the voice itself.
Registration and Readiness
Before a phone can receive calls, it needs to tell the network where it lives. A SIP phone, softphone, or PBX sends a REGISTER request to a registrar, which logs the device's location and authenticates its credentials.
This registration gets refreshed periodically. If it lapses, or if the device is misconfigured or unreachable, inbound calls simply won't arrive.
Several dependencies have to line up for registration to succeed:
- Valid account credentials
- Correct DNS or server settings
- Firewall rules that allow SIP traffic
- Working NAT traversal
- A stable network connection
Skip any one of these and the endpoint may show as registered while still failing to receive calls.
Call Initiation and Routing
Once registered, placing a call starts with an INVITE message. It contains the destination and session details, usually including an SDP offer describing which codecs, ports, and media capabilities the caller supports.
The SIP server, hosted PBX, or provider examines that INVITE and routes it toward the right extension, phone number, call queue, or external carrier network.
As the call progresses, the caller sees a predictable sequence of responses:
| Response Code | What It Means |
|---|---|
| 100 Trying | The server received the request and is working on it |
| 180 Ringing | The destination is alerting the user |
| 183 Session Progress | Call progress information, often with early media |
| 200 OK | The call has been answered |
Authentication challenges, number normalization, busy signals, or routing rules can all alter this basic sequence. A misdialed extension or a busy queue changes what happens next, but the underlying message flow stays consistent.
Negotiation and Session Establishment
After the 200 OK, the receiving endpoint has already returned its own SDP answer, listing its preferred codecs, IP addresses, and ports. Both sides compare notes and settle on compatible parameters. This exchange is formalized in RFC 3264's offer/answer model.
The caller then sends an ACK, confirming receipt of the successful response. At that point, the session is officially established.
SIP messages manage the conversation's state, but audio doesn't travel inside them. Codec mismatches, limited bandwidth, latency, jitter, and packet loss all shape how the audio session sounds.
According to ITU-T Recommendation G.114, keeping one-way delay under 150 milliseconds delivers transparent interactivity for most voice applications. Anything past 400 milliseconds starts feeling noticeably off.
Voice Media Exchange
Once negotiation wraps up, RTP takes over and carries the actual voice packets, either directly between endpoints or through a media relay.
This separation between signaling and media explains a strange but common scenario: a call stays connected on screen while audio goes one-way or disappears entirely. SIP finished its job successfully. RTP is struggling.
Common culprits include:
- NAT or firewall misconfiguration
- Incorrect port handling
- Incompatible codecs
- Poorly tuned session border controllers
RTCP runs alongside RTP, reporting packet loss, jitter, and timing data. Administrators use this feedback to pinpoint whether a quality complaint traces back to the network, the endpoint, or the provider's infrastructure.
A practical example: an employee dials out from a softphone. The call travels through the hosted PBX, exits via a SIP trunk, and lands on an external telephone number. Signaling handles the setup across every hop; RTP carries the actual conversation, sometimes routed through the provider's media servers rather than a direct point-to-point path.
Call Modification, Control, and Termination
SIP doesn't stop working once a call connects. It also manages hold, resume, transfer, conferencing, forwarding, and renegotiation of media parameters mid-call.
Ending the call is handled by a BYE request. The receiving side confirms with a response, and both sides release the session resources.
Abnormal terminations usually trace back to:
- Dropped internet connectivity
- Endpoint or device failure
- Registration expiry
- Provider-side routing issues
- Firewall session timeouts
Why do calls sometimes drop at the 30-second mark? This isn't random. It usually traces to one of these signaling or path failures:
- A missing ACK that leaves the session incomplete
- A NAT or firewall timeout closing an idle media port
- Mishandled SIP responses
- Asymmetric signaling paths (messages take different routes each direction)
That pattern differs from a poor-quality call caused by bandwidth or jitter. It is a signaling or network-path failure, and it usually shows up consistently rather than intermittently.
Where Are SIP Calls Used?
SIP shows up across nearly every business communication workflow:
- Reception desks and auto attendants
- Sales and support call queues
- Internal extensions between departments
- Call recording and conferencing
- Remote work and multi-location calling
It also supports a wide range of endpoints:
- Desk phones and wireless extensions
- Desktop softphones and mobile apps
- Existing IP-PBX systems
- Hosted cloud phone platforms
How those endpoints connect depends on the deployment model you choose.
Hosted PBX vs. SIP Trunking for an Existing System
Businesses generally choose between two paths:
- Hosted/cloud PBX: less IT overhead, no on-site hardware, faster deployment
- SIP trunking with an existing PBX: more control when you already own equipment and want to keep it
The right choice depends on internal IT resources, number portability needs, emergency-calling requirements, redundancy expectations, and how much control the business wants to retain.
Public Telephone Company provides cloud-hosted phone systems, PBX, unified communications, and SIP trunking for businesses that need to scale without buying new hardware.
Its systems run over an existing high-speed internet connection, include equipment with the service instead of an upfront purchase, and support deployments from a handful of users up to 100,000 and beyond. Features and pricing are matched to each deployment.
Pre-Purchase Checklist
Before choosing a SIP-based service, a US business should confirm:
- Internet capacity and QoS support for concurrent calls
- Firewall and NAT readiness for SIP traffic
- Number portability and emergency-calling (E911) compliance
- Security controls and failover options
- Support availability, ideally 24/7
- Any applicable regulatory or telecom tax obligations
Conclusion
SIP signaling locates endpoints, negotiates the session, and ends the call. RTP, running separately, carries the voice itself. That split is the entire logic of how a SIP call works.
Understanding this flow helps businesses diagnose dropped calls, compare hosted PBX against SIP trunking, plan networks that won't choke under load, and pick a provider with the support to back it up.
If a phone system upgrade is on the table, evaluate a scalable cloud-hosted option like Public Telephone Company's PBX, UCaaS, and SIP trunking services against your users, endpoints, network readiness, and existing infrastructure.
Frequently Asked Questions
Is SIP calling free?
SIP-to-SIP calls between compatible apps or endpoints can be free in some setups. Calls to mobile numbers, landlines, or through provider services typically involve charges. Check your provider's rate sheet and plan terms before assuming otherwise.
What is a SIP call?
A SIP call is an IP-based communication session controlled by Session Initiation Protocol, which handles setup, management, and termination. The actual voice audio is carried separately by RTP once SIP finishes negotiating the session.
Why do SIP calls disconnect after 30 seconds?
Common causes include a missing ACK, NAT or firewall timeouts, mishandled SIP responses, or an asymmetric signaling path. SIP traces, endpoint settings, and provider logs usually reveal which one it is.
