SOLUTIONS

SIP Trunk for Enterprise

DID 02, Ribbon SBC, and direct carrier interconnection for enterprise-grade voice quality on a dedicated voice path.

PURE VOIP NETWORK

No reselling. No mixed traffic.

Enterprise-grade SIP Trunk on a dedicated voice path. Not reselling. Thai DID numbers including Bangkok 02 and provincial codes, Ribbon SBC, and direct interconnection to NT and True.

SIPPER runs SIP Trunk on our own Pure VoIP Network, a voice-only lane connected directly to NT (National Telecom) and True Corporation, with no data, video, or other services competing for bandwidth.

Overview

How SIPPER SIP Trunk is different

Most SIP Trunk providers in Thailand resell capacity from carriers like NT (National Telecom) or True Corporation. SIPPER runs SIP Trunk on a Pure VoIP Network, a voice-only lane connected directly to NT and True, with no data, video, or other services competing for bandwidth.

Why 01 / Network

Pure VoIP Network

A voice-only network with no traffic mixing. Voice packets get priority because nothing else is on the line, not just QoS on a shared pipe. Direct interconnection to NT (National Telecom) and True Corporation.

Why 02 / Quality

Ribbon SBC at the Edge

Enterprise-grade Ribbon SBC SWe Edge supporting up to 2,000 concurrent calls with 1:1 active-standby HA and mid-call failover. Microsoft-certified for Teams Direct Routing, Zoom Phone BYOC, and Webex Local Gateway.

Why 03 / Mobile

Mobile SIP Trunk

Use mobile numbers as outbound caller ID (Flexible CLI / CLIP No Screening). Higher answer rates for sales and outbound teams, recipients tend to answer mobile numbers more readily than landlines.

Key Features

Enterprise-grade voice connectivity

A complete feature set for enterprise SIP Trunk, from Thai DID numbers to multi-platform integration.

Thai DID Numbers

DID numbers in Bangkok and provincial regions. Port existing numbers or activate new ones. Geographic, toll-free, and mobile numbers via direct interconnection to carriers NT and True.

Concurrent Channel Model

Pay per concurrent call, not per user or per minute. Add channels instantly without installing physical lines. Scale up or down on demand.

Bundled Calls and Buffet Plans

SIP Trunk packages include free domestic calling. The 12-hour Buffet plan offers unlimited concurrent calls during business hours (choose 6:00-18:00, 7:00-19:00, or 8:00-20:00). Designed for outbound contact centers and sales teams.

TLS/SRTP Encryption

TLS-encrypted signaling and SRTP-encrypted media via SDES key exchange. Ribbon SBC provides topology hiding, DoS/DDoS protection, rate limiting, and rogue RTP filtering.

Flexible Caller ID (CLI)

Display office, mobile, or customer numbers as outbound caller ID. Customer CLI shows the customer's number on outbound calls. Mobile CLI lets PBX users display their mobile number, higher answer rates for sales teams.

Multi-Platform Support

A single SIP Trunk works with Yeastar Cloud PBX, 3CX, Microsoft Teams (Direct Routing), Zoom Phone (BYOC), Asterisk, FreePBX, Cisco, Avaya, and any standard SIP platform. No separate contracts per platform.

Who is it for

SIP Trunk for every voice scenario

Whether you use PBX, Microsoft Teams, or a contact center, SIPPER SIP Trunk provides the voice connectivity layer your platform needs.

Cloud PBX customers

Pair with Yeastar or 3CX Cloud PBX on the Pure VoIP Network. One provider for both PBX and trunk, single point of contact for support and billing.

Microsoft Teams users

Connect Teams Phone to PSTN via Direct Routing. Keep your existing numbers, choose your own carrier, and avoid per-user Calling Plans charges.

On-premise PBX and legacy systems

Replace ISDN PRI/BRI or analog lines with SIP. Compatible with any standard SIP-capable PBX, Yeastar, 3CX, Asterisk, FreePBX, Cisco, Avaya, and others.

All Capabilities

Complete SIP Trunk feature set

Network & Carrier

  • Pure VoIP Network (no traffic mixing)
  • Ribbon SBC SWe Edge (Active-Standby HA)
  • Mid-call Failover (no dropped calls)
  • Direct NT & True Carrier Interconnection
  • Multi-carrier Failover
  • 99.99% SLA
  • 24/7 NOC Monitoring
  • CDR & Call Analytics
  • T.38 Fax over IP
  • DTMF (RFC 2833 / In-band)

Numbers & Calling

  • Thai DID Numbers (Bangkok & provincial)
  • Mobile SIP Trunk (mobile number as CLI)
  • Customer CLI (display customer number)
  • Mobile CLI (display mobile number)
  • Number Porting
  • Concurrent Channel Licensing
  • Bundled Calling Credit in Packages
  • Buffet 12hr Unlimited Calling
  • E.164 Number Formatting
  • G.711 / G.722 / G.729 / Opus Codecs

Integration & Security

  • Microsoft Teams Direct Routing
  • Zoom Phone BYOC
  • Webex Local Gateway
  • Yeastar P-Series Integration
  • 3CX Integration
  • Asterisk / FreePBX Compatible
  • Cisco / Avaya Compatible
  • TLS Signaling + SRTP Media Encryption
  • Topology Hiding (B2BUA)
  • DoS/DDoS Protection
How It Works

Connect your platform to SIPPER SIP Trunk

SIPPER SIP Trunk works with any SIP-capable platform. See how it integrates with the most common setups.

Cloud PBX (Yeastar / 3CX)

When you choose SIPPER Cloud PBX, SIP Trunk is included. The PBX connects to our Pure VoIP Network internally, no public internet for voice traffic between PBX and trunk. Calling packages with bundled minutes available for the highest quality.

Microsoft Teams Direct Routing

SIPPER's Ribbon SBC (Microsoft-certified) connects Teams Phone to SIP Trunk. Calls flow through the Pure VoIP Network to NT and True carriers. Keep your existing numbers, avoid per-user Calling Plan fees, and get enterprise-grade voice quality with mid-call failover.

Mobile SIP Trunk

Use mobile numbers as outbound caller ID on SIP Trunk lines. Packages of 5, 10, 20, or 30 mobile numbers with concurrent channels. Bundled calling credit included or add the 12-hour Buffet plan for unlimited business-hour calling. Designed for outbound sales teams and contact centers where mobile caller ID improves answer rates.

On-premise PBX or Third-party

Connect any SIP-capable PBX to SIPPER SIP Trunk via dedicated or internet connection. We provide SIP credentials, codec configuration (G.711, G.722, G.729, Opus), and firewall guidance. Compatible with Yeastar Appliance, 3CX, Asterisk, FreePBX, Cisco, Avaya, Zoom Phone (BYOC), and Webex.

FAQ

Frequently asked questions about SIPPER SIP Trunk

A network that carries voice traffic only. Unlike typical SIP Trunk providers that send voice over shared internet, the SIPPER Pure VoIP Network carries voice packets only, no data, video, or other services. Direct interconnection to NT (National Telecom) and True Corporation means no bandwidth competition and consistent call quality.

Mobile SIP Trunk lets you use a mobile number as outbound caller ID when calling through your PBX or contact center. Using Flexible CLI (CLIP No Screening) technique, the mobile number is displayed instead of the landline. Available in packages of 5 to 30 mobile numbers with bundled calling credit, or add the 12-hour Buffet plan for unlimited calling.

The Buffet plan provides unlimited concurrent calls during a 12-hour business-hours window (choose 6:00-18:00, 7:00-19:00, or 8:00-20:00). Available for both DID SIP Trunk (1 to 50 concurrent channels) and Mobile SIP (5 to 30 numbers). Designed for outbound contact centers and high-volume sales teams during business hours.

Yes. SIPPER manages the entire number porting process end-to-end. We coordinate with your existing provider and the carrier (NT or True) to transfer DID numbers with minimal downtime. Most ports complete within 5-10 business days.

SIPPER SIP Trunk uses a concurrent channel model, packages from 1 to 50+ concurrent calls. The Ribbon SBC SWe Edge supports up to 2,000 concurrent sessions. Add or remove channels as call volume changes, no physical line installation required.

Yes. The same SIP Trunk supports Microsoft Teams Direct Routing, Zoom Phone BYOC, and Webex Local Gateway via Ribbon SBC certified by Microsoft, Zoom, and Cisco. One trunk, one provider, one number set for every platform.

SIPPER uses 1:1 active-standby Ribbon SBC pairs with mid-call failover, active calls stay connected even during hardware issues. Multi-carrier failover automatically routes calls through backup carriers. Combined with DoS/DDoS protection and 24/7 NOC monitoring for a 99.99% SLA.

SIPPER SIP Trunk supports G.711 (ulaw/alaw) for standard voice, G.722 for HD wideband voice, G.729 for bandwidth efficiency, and Opus for adaptive quality. Codec negotiation is automatic via Ribbon SBC with GPU-accelerated transcoding. T.38 is supported for fax over IP.

Connect your business to a better voice network

Tell us about your current phone system, call volume, and requirements. We'll design a SIP Trunk that fits your platform and budget.

Sipper
Sipper Network Communications Co., Ltd. (Head Office)

Tax ID: 0105560159831

99/4 New Connex House, Don Mueang, Phaholyothin Road, Sanambin, Don Mueang, Bangkok 10210
02-098-9500
Hotline Support (24 Hours.): 02-666-9494
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02-098-9500
02-666-9494
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