Standard SIP protocol interconnection, compatible with mainstream PBX and softswitch platforms, elastic concurrent channel scaling, and fast access to voice networks across 200+ countries
Free TrialKey parameters for buyers and technical teams evaluating PBX, call center, and global voice connectivity.
Asterisk, FreeSWITCH, 3CX, Cisco CUBE, Avaya and other PBX or softswitch integrations, plus outbound call center concurrency expansion.
Standard SIP/RTP, IP whitelist or SIP registration authentication, UDP/TCP/TLS options, and common enterprise voice codecs.
10 to 10,000+ concurrent channels, 200+ countries and regions, with core strengths in North America, Europe, and Southeast Asia.
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Full-featured voice trunking based on standard SIP/RTP protocols
Fully compliant with SIP RFC 3261 and RTP RFC 3550 standards, ensuring interoperability with all mainstream communication equipment. Supports SIP over UDP/TCP/TLS.
Scale concurrent channels on demand without pre-purchasing fixed capacity. From 10 to 1,000+ channels with minute-level elastic adjustment, pay only for actual usage.
Supports G.711(A/U), G.729(AB), G.722, Opus and other mainstream codecs with automatic optimal codec negotiation. Supports codec passthrough and transcoding.
Coverage in 200+ countries and regions with 30+ global POP nodes and direct connections to major carriers. North America, Europe, and Southeast Asia are our core strength regions.
Compatible with Asterisk, FreeSWITCH, 3CX, Cisco CUBE, Avaya, Yeastar, Elastix and other mainstream PBX and softswitch platforms.
Supports both IP whitelist authentication (recommended) and SIP registration authentication (username + password) to accommodate different network environments.
Complete technical parameters and compatibility information
| Signaling Protocol | SIP (RFC 3261) over UDP / TCP / TLS |
| Media Protocol | RTP (RFC 3550) / SRTP (RFC 3711) |
| Codecs | G.711 (PCMA/PCMU), G.729 (A/B), G.722, Opus, G.726, iLBC |
| DTMF | RFC 2833, In-band, SIP INFO |
| Fax | T.38 (FoIP), G.711 passthrough |
| NAT Traversal | STUN, TURN, ICE, Media Relay |
| Codec Negotiation | SDP Offer/Answer (RFC 3264) |
| Concurrent Capacity | 10 - 10,000+ channels, elastic scaling |
| SLA | 99.99% availability guarantee |
| Authentication | IP whitelist / SIP registration (Digest auth) |
| PBX Compatibility | Asterisk, FreeSWITCH, 3CX, Cisco CUBE, Avaya, Yeastar, Elastix, Issabel, Brekeke |
| Monitoring | Real-time CDR, call quality monitoring, MOS scoring, anomaly alerts |
SIP trunking empowers a variety of enterprise communication scenarios
Unified access for multi-country DID numbers with centralized agent management. Supports IVR, queuing, recording and other call center features. Quickly provision international customer service lines via SIP trunking.
Connect PBX systems across global offices with free internal calls and unified routing and billing for external calls. Enable unified communications for multinational enterprises.
High-concurrency voice verification code and notification outbound calls, triggered via API. Ideal for large-scale voice delivery needs of internet businesses.
Direct SIP trunking connection with Microsoft Teams, Zoom Phone, Cisco WebEx and other UC platforms, providing enterprises with a complete voice solution.
Learn more about SIP trunking technology