Solutions
Enterprise-grade SIP Trunk running on SIPPER's own voice network — no mixed traffic, no reselling. Thai DID numbers, Ribbon SBC at the edge, and direct carrier interconnection for consistent call quality.
Overview
Most SIP Trunk providers in Thailand resell capacity from carriers like NT (National Telecom) or True Corporation. SIPPER operates its own Pure VoIP Network with direct carrier interconnection — dedicated voice infrastructure with no data, video, or other services competing for bandwidth.
Dedicated voice-only network with no mixed traffic. Voice packets get priority because there is nothing else on the wire — not QoS tagging on a shared pipe. Direct interconnection with NT (National Telecom) and True Corporation carriers.
Enterprise-grade Ribbon SBC SWe Edge — supports up to 2,000 concurrent sessions with 1:1 active-standby HA and mid-call failover. Microsoft-certified for Teams Direct Routing, Zoom Phone BYOC, and Webex Local Gateway.
Present mobile phone numbers as outbound caller ID (Flexible CLI / CLIP No Screening). Higher answer rates for sales and outbound teams — recipients are more likely to pick up calls from mobile numbers than landlines.
Key Features
Local DID numbers in Bangkok and regional areas. Port existing numbers or provision new ones. Support for geographic, toll-free, and mobile numbers via NT and True carrier interconnection.
Pay for the number of simultaneous calls you need, not per-user or per-minute. Scales instantly without physical line installation — add or remove channels by agreement.
SIP Trunk packages include free domestic calling credit. Buffet 12-hour plans offer unlimited concurrent calls during business hours (6:00–18:00, 7:00–19:00, or 8:00–20:00) — ideal for outbound contact centers and sales teams.
Signaling encrypted with TLS, media encrypted with SRTP via SDES key exchange. Ribbon SBC provides topology hiding, DoS/DDoS protection, rate limiting, and rogue RTP filtering.
Present landline, mobile, or customer numbers as outbound caller ID. Customer CLI shows your customer's number on outgoing calls. Mobile CLI lets PBX users show their mobile number — higher answer rates for sales.
One SIP Trunk for Yeastar Cloud PBX, 3CX, Microsoft Teams (Direct Routing), Zoom Phone (BYOC), Asterisk, FreePBX, Cisco, Avaya, and any SIP-compliant platform. No separate contracts per platform.
Who is this for
Whether you run a PBX, use Microsoft Teams, or operate a contact center — SIPPER SIP Trunk provides the voice connectivity layer your platform needs.
Bundled with Yeastar or 3CX Cloud PBX on SIPPER's Pure VoIP Network. One provider for both PBX and trunk — single point of support and billing.
Connect Teams Phone to PSTN via Direct Routing. Keep your numbers, choose your carrier, and avoid Microsoft Calling Plans per-user fees.
Replace ISDN PRI/BRI or analog lines with SIP. Works with any standards-based PBX — Yeastar, 3CX, Asterisk, FreePBX, Cisco, Avaya, and more.
Full Capabilities
How It Works
SIPPER SIP Trunk connects to any SIP-compatible platform. Here is how it works with the most common setups.
When you choose SIPPER Cloud PBX, SIP Trunk is already bundled. Your PBX connects to our Pure VoIP Network internally — no public internet for voice traffic between PBX and trunk. Calling plans with included credit available. This is the highest-quality deployment option.
SIPPER's Ribbon SBC (Microsoft-certified) connects your Teams Phone system to our SIP Trunk. Calls route through our Pure VoIP Network to NT and True carriers. You keep your existing phone numbers, avoid per-user Calling Plan fees, and get enterprise-grade voice quality with mid-call failover.
Use mobile phone numbers as outbound caller ID on SIP Trunk calls. Available in packages of 5, 10, 20, or 30 mobile numbers with matching concurrent channels. Bundled calling credit included, or add Buffet 12-hour plans for unlimited calling during business hours. Ideal for outbound sales and contact centers where mobile caller ID increases answer rates.
Connect any SIP-compatible PBX to SIPPER SIP Trunk over a dedicated or internet connection. We provide SIP credentials, codec configuration (G.711, G.722, G.729, Opus), and firewall guidance. Works with Yeastar Appliance, 3CX, Asterisk, FreePBX, Cisco, Avaya, Zoom Phone (BYOC), and Webex.
FAQ
A network dedicated exclusively to voice traffic. Unlike typical ISP-based SIP Trunk providers who run voice over shared internet infrastructure, SIPPER's Pure VoIP Network carries only voice packets — no data, video, or other services. Direct interconnection with NT (National Telecom) and True Corporation eliminates bandwidth contention and delivers consistent call quality.
Mobile SIP Trunk lets you present mobile phone numbers as outbound caller ID when calling through your PBX or contact center system. This uses Flexible CLI (CLIP No Screening) to show a mobile number instead of a landline to called parties. Available in packages of 5 to 30 mobile numbers with bundled calling credit or Buffet 12-hour unlimited calling plans.
Buffet plans provide unlimited concurrent calls during a 12-hour business window (choose 6:00-18:00, 7:00-19:00, or 8:00-20:00). Available for DID SIP Trunk (1 to 50 concurrent channels) and Mobile SIP (5 to 30 numbers). Ideal for outbound contact centers and sales teams that make high volumes of calls during working hours.
Yes. SIPPER handles the number porting process end-to-end. We coordinate with your current provider and the carrier (NT or True) to transfer your 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+ simultaneous calls. Our Ribbon SBC SWe Edge supports up to 2,000 concurrent sessions. Channels can be increased or decreased as your call volume changes — no physical line installation required.
Yes. The same SIP Trunk powers Microsoft Teams Direct Routing, Zoom Phone BYOC, and Webex Local Gateway through our Ribbon SBC — which is certified by Microsoft, Zoom, and Cisco. One trunk, one provider, and one set of phone numbers across all platforms.
SIPPER runs a 1:1 active-standby Ribbon SBC pair with mid-call failover — active calls are preserved even during hardware failure. Multi-carrier failover automatically routes calls through an alternate carrier path. Combined with DoS/DDoS protection and 24/7 NOC monitoring, this delivers our 99.99% SLA.
SIPPER SIP Trunk supports G.711 (ulaw/alaw) for standard voice, G.722 for wideband HD audio, G.729 for bandwidth-efficient calls, and Opus for adaptive quality. Codec negotiation is automatic through our Ribbon SBC with GPU-accelerated transcoding. T.38 is supported for fax over IP.
Get Started
Tell us about your current phone system, call volume, and requirements. We will design a SIP Trunk solution that fits your platform and budget.