Recommended for security-sensitive industries: finance, healthcare, or organizations needing PDPA / GDPR compliance. SRTP encrypts media so eavesdroppers cannot decode the audio. TLS encrypts signaling.

Yes, common causes include internet outage, ISP problems, provider SBC failures, or DDoS attacks. A robust design uses dual internet links, trunks from two different carriers, and redundant SBCs to keep the phone system online.

What Is SIP Trunk? Complete 2026 Guide

Q: How does SIP Trunk differ from Cloud PBX

SIP Trunk is the voice path connecting a customer PBX to the public phone network. Cloud PBX is the phone system itself, hosted on the provider's cloud. Customers with their own PBX (like an on-site Yeastar P-Series) buy SIP Trunks for outbound calling. Customers using Cloud PBX usually get SIP Trunks bundled into their package.

Q: Can I use regular internet for SIP Trunk

Technically yes, but not recommended for production. Regular internet has no QoS, leading to inconsistent voice quality. Enterprise providers typically offer a dedicated pure VoIP network or MPLS connection guaranteeing latency, jitter, and packet loss.

Q: How many channels do I need

Calculate from peak-hour traffic using an Erlang B calculator. As a guideline, a 50-person office may need 5 to 10 channels, while a 30-agent contact center needs 25 to 35. Quality providers help size based on your existing CDR data.

Q: Does SIP Trunk support Microsoft Teams

Yes, via Direct Routing. SIP Trunks can connect to Microsoft Teams through a certified SBC (AudioCodes, Ribbon, Oracle, Sangoma) so Teams users can place PSTN calls at lower cost than Microsoft Calling Plans.

Q: Can I port my existing 02-XXX-XXXX number to a SIP Trunk

Yes, through number porting. The new SIP Trunk provider coordinates with your old carrier (NT, True) to transfer numbers. Process takes 5 to 15 business days with very short downtime on the cutover day.

2026-05-07 9 min readSIPPER Team

SIP Trunk (Session Initiation Protocol Trunk) is a service that connects your phone system to the public switched telephone network (PSTN) over IP, replacing copper analog trunks or ISDN E1 / PRI lines. This guide covers definition, architecture, comparison with ISDN, key features, pricing, and how to choose a SIP Trunk provider in Thailand.

What Is SIP Trunk

A SIP Trunk is a virtual voice "trunk" that carries voice calls between your PBX and a telecom carrier using the SIP protocol over an IP network, replacing legacy copper analog trunks or digital ISDN E1 / PRI lines.

Technically, SIP Trunks send signaling (call control) over SIP and media (audio) over RTP / SRTP. This lets your phone system place and receive external calls just like a traditional trunk, but using IP bandwidth instead of physical lines.

SIP Trunks are ideal for organizations that already have an IP PBX or Cloud PBX and need to connect to the public telephone network for inbound and outbound calls.

How SIP Trunk Works

A SIP Trunk has three core components: registration, call setup, and media stream.

Registration

The customer PBX registers with the provider's SBC using SIP username, password, and domain to declare it is online. There are two main forms: Registered Trunk (the PBX sends periodic REGISTER messages) and IP-based Trunk (authentication by the customer's public IP, no REGISTER required).

Call Setup

For inbound calls from PSTN, the carrier sends a SIP INVITE to the PBX. The PBX replies with 100 Trying / 180 Ringing / 200 OK. For outbound calls, the customer PBX sends INVITE to the carrier, which then routes the call out to PSTN.

Media Stream

After the call connects, audio flows over RTP (or SRTP for encryption) between the PBX and the provider SBC. Common codecs include G.711 (ulaw / alaw), G.722 (HD voice), G.729 (compressed), and Opus.

How SIP Trunk Differs From ISDN E1 / PRI

ISDN E1 / PRI is a legacy digital telephony line that delivers 30 channels per circuit using copper wiring and dedicated hardware. SIP Trunk uses IP networking, making it far easier to scale and manage.

Connectivity: ISDN uses copper wires / SIP Trunk uses internet or pure VoIP networks
Channels: ISDN E1 = fixed 30 channels / SIP Trunk scales up or down on demand
Cost: ISDN has high monthly rental + per-minute / SIP Trunk is cheaper monthly + lower per-minute
Hardware: ISDN needs legacy PBX + E1 card / SIP Trunk uses IP PBX or Cloud PBX through an SBC
Voice quality: ISDN limited to G.711 / SIP Trunk supports HD voice (G.722, Opus)
Flexibility: ISDN tied to physical site / SIP Trunk can move with the server

Key Components of a SIP Trunk

A real production SIP Trunk deployment involves more than just turning on the service.

Session Border Controller (SBC)

The SBC sits at the border between the carrier and the customer network, handling security, NAT traversal, codec transcoding, topology hiding, and survivability. Some organizations run their own SBC, others rely on a hosted SBC from the provider.

Concurrent Channels

The number of simultaneous calls a trunk supports. Sizing should be based on peak-hour traffic, often calculated with Erlang B. A 50-agent contact center may need 30 concurrent channels. Trunks are typically sold in packs of 5 / 10 / 20 / 50+.

DID (Direct Inward Dial)

Phone numbers attached to the trunk. Customers can buy multiple DIDs per trunk: Bangkok (02) numbers, provincial numbers, or 1800 / 1900 / toll-free numbers.

Failover and Redundancy

Best practice is to have a backup trunk (active-active or active-standby) on a different carrier to avoid a single point of failure on either internet path or voice network.

Enterprise-Grade SIP Trunk Features

Enterprise SIP Trunks should offer more than just basic call delivery.

HD voice (G.722, Opus) for crystal-clear audio
SRTP / TLS for media and signaling encryption
CLIP / CLIR (caller ID display and suppression)
Provider-side call recording for compliance
CDR (Call Detail Record) export for cost analysis
Number porting to migrate existing numbers
Concurrent channel burst for marketing campaigns
Self-service portal for routing changes, usage, CDR download

Typical SIP Trunk Pricing in Thailand

SIP Trunk pricing in Thailand depends on the carrier, channel count, and call package selected.

Monthly rental starts around 500 to 1,500 THB per channel. Domestic calls are typically 0.50 to 1.50 THB per minute, with international rates varying by destination.

Many providers offer bundles with unlimited domestic minutes for 2,000 to 5,000 THB per channel per month, ideal for contact centers or businesses with high outbound volume.

One-time activation fees range from 0 to 10,000 THB per trunk depending on promotion and trunk size.

How to Choose a SIP Trunk Provider

Beyond price, evaluate network quality and post-sales support.

Carrier interconnection: direct with NT, True, AIS or reseller layered
Pure VoIP network: dedicated voice path or shared internet
PDPA: CDR and recordings stored in Thailand or abroad
SLA: 99.9 vs 99.99 percent uptime guarantee
Support: 24x7 NOC with Thai-language coverage
PBX compatibility: Yeastar, 3CX, Asterisk, Avaya, Cisco
Self-service portal: change configs, view usage independently

Frequently Asked Questions

How does SIP Trunk differ from Cloud PBX

Can I use regular internet for SIP Trunk

How many channels do I need

Does SIP Trunk support Microsoft Teams

Is SRTP necessary

Can I port my existing 02-XXX-XXXX number to a SIP Trunk

Can SIP Trunks fail

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