Direct SIP Termination
Low-Latency RFC 3261 SIP Peering for Voice Aggregators
Direct SIP termination engineered specifically for VoIP platforms, CPaaS providers, and enterprise cloud PBX environments. Deliver precise signaling execution, accurate G.711/G.729 codec negotiation, and optional cryptographic media security across global termination legs.
Architectural Capabilities & Features
RFC 3261 SIP Signaling
Rigorous SIP protocol implementation ensuring accurate Session Description Protocol (SDP) negotiation and RFC 2833 dual-tone DTMF relay.
Optimized Connection Latency
Engineered routing paths designed to minimize signaling delays across intermediate carrier tandems for immediate session establishment.
Media Payload Security
End-to-end TLS signaling encryption combined with Secure RTP (SRTP) media encryption for voice payload protection.
Disconnect Cause Code Telemetry
Standard RFC 3326 Reason headers mapping ITU-T Q.850 release cause codes directly to your switch for accurate call failure diagnostics.
Technical Specifications
| Supported Signaling | SIP RFC 3261, UDP, TCP, TLS |
| Supported Codecs | G.711u, G.711a, G.729a, G.722, Opus |
| DTMF Relays | RFC 2833, RFC 4733, SIP INFO, In-band |
| NAT Traversal | Far-end NAT traversal with proactive RTP latching |
| SBC Redundancy | Active-active Session Border Controller clustering |