
If you want to know the details of how SIP trunking works, then I will explain here everything you need to know and the process too.
What Is SIP Trunking?
A SIP trunk is an essential system of virtual bundles of voice channels carried over IP by using SIP. It works for the setup, management, and tear-down of phone calls. Otherwise, traditional copper phone lines aren’t like these. It’s costly, a maintenance hassle, and has weather-related issues. That’s why people are moving to the SIP or internet-based communication system.
Key advantages:
- Scalability: You can add or remove channels on demand.
- Cost Efficiency: pay per channel or minute, no physical lines.
- Feature-Rich: Leverage advanced call routing, disaster recovery, and codecs.
Core Components & Roles
IP-PBX or Hosted PBX
Your on-prem or cloud-hosted telephony platforms, such as Free PBX, Cisco CUCM, 3CX, or our Cloud PBX that manages extension, call routing, and voicemail, etc.
Session Border Controller (SBC)
Acts as a security and interoperability gatekeeper that hides your internal IP scheme, enforces encryption, and applies SIP normalization
Internet Telephony Service Provider (ITSP)
The Voice providers that terminate calls to the PSTN or other SIP networks and allocate you a pool of DIDs with the voice channels.
Media Path (RTP/RTCP)
Carries actual voice packets, typically RTP with RTCP for quality control. It may traverse a separate media proxy or go peer-to-peer via NAT traversable techniques.
Signaling Path, SIP Messages: INVITE, 180 Ringing, 200 OK, ACK, BYE, etc., traveling over UDP/TCP/TLS.

How SIP and VoIP Work Together
Well, VoIP is the technology that businesses use to communicate. As well, SIP is the methods or protocol that execute its technology for terminating voice.
If I want to be more clear with the real thought, I think VoIP is a car and SIP is the engine of this system.
Real-World Example
MondoTalk’s Cloud PBX:
Using VoIP, you can make calls to the customer using broadband. SIP trunks to connect clients’ PBX systems to MondoTalk’s core telecom network. It offers IVR, Call transfer, CRM, MS team integration, Voice to email, mobile apps, and failover routing.
Common Combination Setup:
Component | Technology |
Softphone App | VoIP (with SIP) |
IP Desk Phone | SIP-enabled |
PBX System | SIP-based (Cloud or On-Premise) |
Call Routing & Trunking | SIP Trunks |
Internet Connection | Underlying VoIP Carrier |
How SIP Trunking Works The Process
Let’s break down step by step what happens when you make a VoIP Call through SIP trunking.
1. Call Initiation (SIP Signaling Begins)
The moment you dial a number, your IP PBX generates a SIP INVITE message.
This INVITE packet includes:
- Caller ID
- Callee’s number
- Codec preferences
- Authentication details
Then the message reached the SIP provider such as MondoTalk, Ufonii, or Twilio.
2. SIP Trunk Authentication and Routing
The provider receives your INVITE and authenticates it for use with username/password. This authentication will be IP-based. I mean whitelisted IPs. Once authenticated, the SIP server then checks the destination numbers and chooses the best route.
3. Media Negotiation (RTP Stream Setup)
- SIP only sets up the session, and voice is sent via RTP( Real Time Transport Protocol).
- Once the destination accepts the call, like a SIP 200 OK, both parties start sending voice data as audio packets.
- RTP ports that are typically 10000-20000 open up dynamically, and NAT traversal or STUN/TURN are used if needed.
4. Call Connection and Ongoing Session
- During the call:
- SIP monitors session health via OPTIONS and SIP keep-alives.
- Media streams flow back and forth using RTP.
- SIP monitors session health via OPTIONS and SIP keep-alives.
- Features such as DTMF tones, call recording, or transferring are also handled via in-band RTP or SIP info/notify messages.
5. Call Termination
- Either party hangs up: sends a SIP BYE message.
- Provider replies with 200 OK, confirming the call is terminated.
- Call Detail Records (CDRs) are generated for billing and logs.
Layer | Protocols / Processes |
Signaling | SIP (RFC 3261), with SIP Methods (INVITE, BYE, REGISTER, OPTIONS, etc.) |
Media | RTP (RFC 3550), optionally SRTP for encryption |
NAT Traversal | STUN, TURN, ICE (for peer-to-peer media) |
Security | TLS for SIP, SRTP for media, ACLs on SBC |
Codec Negotiation | SDP offer/answer; common codecs: G.711, G.722, G.729 |
Numbering Plan | E.164 formatting, digit manipulation rules |
Real-World Example
Example: Retail Call Center with Cloud PBX
Company: Local Australian electronics retailer
Setup:
- 1 cloud PBX (3CX)
- SIP trunk
- 30 concurrent calls
How it works daily:
- Customer calls 1300 number: hits Ufonii SIP servers.
- Ufonii routes call their 3CX IP PBX.
- IVR picks up routes to departments.
- Agents use softphones (Zoiper/Bria) or IP phones.
- Voice quality is monitored in real-time via MOS scoring.
- CDRs sync into Zoho CRM for call reports.
Personal Experience Insight:
We once had intermittent audio drops on rainy days. It turned out to be a poor-quality router at the retail site doing SIP ALG rewrites. We replaced it with a proper SIP-friendly route,r and the issue disappeared.
Example: Legal Firm
Helped a mid-size legal firm with 75 extensions replace their aging PRI circuits with a pair of redundant SIP trunks. They’d been paying for 23 channels on two ISDN PRIs.
Preparation Steps:
- Bandwidth Assessment
Run a year’s worth of call logs and peak concurrent calls: 18. It allocates 100 kbps per G.711 call, 1.8 Mbps plus overhead, and 3 Mbps reserved.
- SBC Deployment
Installing an on-premises SBC and configuring TLS on port 5061. It defined IP ACLs so that only the ITSP’s signaling IPs could connect.
- PBX Configuration
You can create two outbound trunks on the PBX:
- SIP-Primary (UDP/5060, priority 1)
- SIP-Backup (TCP/TLS/5061, priority 2): Set up inbound routes mapping each DID to a departmental ring group.
- SIP-Primary (UDP/5060, priority 1)
- Test Calls & Failover
Then you can verify calls over SIP primary to PSTN. The system simulated failure by shutting off the primary, and calls seamlessly failed over to the backup.
- Monitoring & QoS
Then configure SNMP polling on the SBC for call count, SIP errors, and set diffServ DSCP EF for RTP and CS3 for SIP to prioritize voice.
SIP Trunking: Common Technical Components
Component | Role in SIP Trunking |
SIP Proxy | Handles SIP message routing & signaling |
Registrar Server | Authenticates users/IPs |
RTP Engine | Handles actual audio/video streaming |
SBC (Session Border Controller) | Security, NAT traversal, protocol normalization |
Media Gateway | Converts SIP to TDM for PSTN interconnects |
Behind-the-Scenes Security
- SIP over TLS: Encrypts signaling messages (prevents spoofing)
- SRTP: Encrypts RTP audio (prevents call tapping)
- SBCs: Firewall + protocol guard
- Call Limits & IP ACLs: Prevent fraud
Personal Expertise: Lessons from the Field
- Never underestimate the power of good call logs.
If there is a SIP 503 or 408 error, then reading the SIP invite header, RTP latency, and timing in Wireshark has solved countries - Carrier Hops Matter:
A poor SIP provider might route your call through 4-5 unnecessary hops, creating jitter, which reliable providers like Ufonii route more directly. - Don’t DIY NAT Traversal:
Always use proper NAT rules and test with external SIP test tools like sipvicious or VoIPmonitor. - Real Call Center Experience: A major partner once migrated from ISDN to SIP trunking. Then the experience with SIP trunk, such as post migration, and they were able to slash their monthly bill by 40%, and they were able to add auto call routing and enable CRM integration.
Benefits Recap
SIP Trunking Advantage | Explanation |
Scalability | Add/remove call capacity instantly |
Cost Efficiency | No per-line charges like PSTN |
Flexibility | Can be used across geographies with one trunk |
Integration | Easily integrates with CRMs, APIs |
Advanced Features | Call forwarding, recording, transcription, so on etc. |
Testing Tools & Protocols
- Wireshark – Packet capture for SIP, RTP.
- sngrep – CLI SIP session viewer
- VoIP monitor – Call quality analytics
- SIP – SIP stress testing tool
- 3CX / Asterisk / FreePBX – Real PBX platforms to test with
Final Thoughts
SIP trunking is the backbone of the modern communication system.. Behind this scene, there are lots of dependencies such as a symphony of protocols, routing, codecs, and media negotiation. The beauty is in how many moving parts come together to make a simple” hello ”work. I mean it started from INVITE packets to RTP streams and from NAT traversal to carrier selection.
When it works well, it feels like magic.
FAQs
What happens when I make a SIP trunk call?
When you make a SIP trunk call, your phone system sends a SIP invitation packet to your VoIP provider. This includes the dialed number, caller ID, and preferred audio settings. The provider routes the call to the destination, and once answered, media flows directly via RTP. SIP keeps the session alive, monitors connection health, and handles call termination when either party hangs up.
How does SIP trunking connect to the public telephone network (PSTN)?
That’s a great question. Generally, SIP trunking connects to the PSTN via your VoIP provider, which acts as a gateway. Once your call reaches the provider’s SIP servers, it is routed through their carrier network and translated into a traditional phone call that can be handled by legacy telephone systems. It enables the voip to make landline or VoIP to mobile calls seamlessly.
Is SIP trunking secure?
Yes, SIP trunks are secure for businesses. But it requires a proper setup before using. SIP trunking generally uses protocols like SIP over TLS and SRTP to encrypt signaling and media streams. Session border controllers also help protect against fraud, unauthorized access, and DDoS attacks. The firewalls should be configured to avoid exposing SIP ports unnecessarily.
Why is RTP used in SIP trunking, and what is it?
RTP( Real Time Transport Protocol is used to carry the actual voice data once a call is established. While SIP handles the signaling, RTP transmits the audio packets in real time. The separation ensures flexibility and allows for better quality control and troubleshooting.
What are the most common issues with SIP trunking?
The most common SIP trunking issues include:
- One-way or no audio. It usually happens due to NAT/firewall misconfiguration.
- Call drops (caused by SIP timeouts or router issues)
- Codec mismatches
- SIP ALG interference on routers
How many calls can a SIP trunk handle?
A single SIP trunk can support multiple concurrent calls, it’s depending on your provider and bandwidth. Each simultaneous call typically requires around 100 Kbps with the G.711 codec, so your internet connection and trunk plan determine the actual capacity. Businesses can scale easily by increasing call channels.
What equipment is needed for SIP trunking?
To use SIP trunking, you typically need
- IP PBX such as 3CX, FreePBX, and Cloud-based PBX
- A router and a stable internet connection
- SIP trunk provider credentials
- VoIP phones, softphones, or analog phones with ATA adapters
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