🤖SIP Interoperability Guide
Trusst AI Voice Platform Technical Integration Guide for Telecommunications Providers
1. Document Purpose
This document provides telecommunications providers with the technical specifications and configuration requirements necessary to establish SIP trunk connectivity with the Trusst AI Voice Platform. The integration enables AI-powered voice agents to handle inbound and outbound telephone calls through your carrier infrastructure.
Trusst's platform supports real-time AI voice interactions for contact center automation, customer service, appointment scheduling, and various other telephony-based AI applications. This guide covers the complete configuration workflow for establishing bidirectional SIP trunk connectivity.
2. Platform Overview
2.1 Architecture Summary
The Trusst AI Voice Platform operates as a cloud-based SIP endpoint that bridges traditional PSTN telephony with real-time AI voice processing. The platform accepts SIP INVITE requests for inbound calls and originates SIP sessions for outbound calls through configured trunk connections.
Key architectural components include:
SIP signaling gateway for call control
Media processing infrastructure for real-time audio
AI voice agents for natural language interactions
Call routing logic for dispatch and transfer operations
2.2 Supported Use Cases
Inbound AI Agents
Receive incoming calls and route to AI-powered voice agents for automated handling
Outbound AI Campaigns
Initiate outbound calls from AI agents for proactive customer engagement
Hybrid Operations
Support for both inbound and outbound calling through a unified trunk configuration
Call Transfer
AI agents can transfer calls to external numbers or contact center agents via SIP REFER or re-INVITE
3. Technical Requirements
3.1 SIP Protocol Specifications
SIP Version
SIP/2.0 (RFC 3261)
Transport Protocols
UDP, TCP, TLS
SIP Signaling Port
5060 (UDP/TCP), 5061 (TLS)
Authentication
Digest Authentication (username/password) or IP-based ACL
Session Timers
Supported (RFC 4028)
DTMF Method
RFC 2833 (telephone-event) – Required
3.2 Audio Codec Requirements
The Trusst platform supports standard telephony codecs. Codec negotiation follows SDP offer/answer model (RFC 3264). The following codecs are supported in order of preference:
G.711 µ-law (PCMU)
0
8 kHz
64 kbps
G.711 A-law (PCMA)
8
8 kHz
64 kbps
G.722
9
16 kHz
64 kbps
telephone-event
101 (dynamic)
8 kHz
N/A
3.3 Media (RTP) Specifications
RTP Port Range
10000–20000 (dynamic allocation)
Media Encryption
SRTP supported (AES-128-CM); optional based on configuration
Packetization Time
20ms (ptime=20)
SDP Protocol
RTP/AVP (unencrypted) or RTP/SAVP (encrypted)
IP Version
IPv4 (IPv6 available upon request)
3.4 Network Requirements
To ensure optimal voice quality for AI interactions, the following network parameters must be maintained:
Latency
< 150ms one-way (< 100ms recommended for real-time AI responsiveness)
Jitter
< 30ms
Packet Loss
< 1%
Bandwidth per Call
~100 kbps (G.711 with overhead)
4. Inbound Trunk Configuration
Configure inbound trunking to route incoming calls from your network to the Trusst platform. This enables AI agents to answer and process inbound customer calls.
4.1 Trusst SIP Endpoint
Direct SIP INVITE requests for inbound calls to the Trusst SIP endpoint. Your dedicated SIP URI will be provided during onboarding in the following format:
sip:{your-tenant-id}.sip.trusst.cloudIf your carrier configuration requires an endpoint format without the sip: prefix, use:
{your-tenant-id}.sip.trusst.cloud4.2 Configuration Steps
Configure Origination URI: Set the Trusst SIP endpoint as the origination destination for the phone numbers designated for AI handling.
Set Transport Protocol: Configure TCP or TLS transport. TLS (port 5061) is recommended for production environments.
Configure Number Format: Set destination number format to E.164 with leading '+' (e.g.,
+15105550100).Associate Phone Numbers: Assign the purchased DID numbers to the configured SIP connection/trunk.
Verify Routing: Confirm that inbound calls to the designated numbers are routed to the Trusst endpoint.
5. Outbound Trunk Configuration
Configure outbound trunking to enable the Trusst platform to initiate calls through your carrier network. This is required for outbound AI campaigns, callbacks, and call transfers.
5.1 Authentication Configuration
Trusst supports digest authentication (username/password) for outbound trunk connectivity. Provide the following credentials during trunk provisioning:
SIP Domain/Address
Your carrier's SIP proxy FQDN (e.g., sip.carrier.com)
Auth Username
SIP digest authentication username
Auth Password
SIP digest authentication password
Outbound Numbers
E.164 formatted numbers authorized for caller ID
Transport
UDP, TCP, or TLS
5.2 IP Allowlisting (If Required)
If your carrier requires IP-based authorization for outbound calls, note that Trusst operates from dynamically allocated cloud infrastructure. We recommend using digest authentication rather than IP allowlisting.
If IP allowlisting is mandatory, please contact Trusst support for current egress IP ranges for your deployment region.
5.3 Number Format Requirements
Ensure your carrier is configured to accept the following number formats from Trusst:
Caller ID (From header)
E.164 format with leading '+' (e.g., +15105550100)
Destination (To header)
E.164 format with leading '+'
Request URI
sip:+{number}@{carrier-domain}
6. Security Considerations
6.1 Encryption Recommendations
For production deployments handling sensitive customer interactions, we recommend:
TLS 1.2 or TLS 1.3 for SIP signaling encryption
SRTP (Secure RTP) for media encryption using AES-128
Digest Authentication rather than IP-only authorization
6.2 Fraud Prevention
Implement standard telephony fraud prevention measures including:
Call rate limiting
Geographic restrictions where appropriate
Caller ID validation
Monitoring for anomalous call patterns
Trusst implements platform-level protections, but carrier-side controls provide defense in depth.
7. Testing and Validation
7.1 Pre-Production Checklist
Complete the following validation steps before enabling production traffic:
Inbound call connects and audio path established
☐
Outbound call connects with correct caller ID
☐
Bidirectional audio confirmed (no one-way audio)
☐
DTMF tones detected correctly (RFC 2833)
☐
Call hangup terminates session cleanly (BYE processed)
☐
Codec negotiation successful
☐
Call transfer completes successfully (if applicable)
☐
TLS/SRTP encryption verified (if enabled)
☐
8. Support Contact
For technical assistance with SIP trunk integration, contact Trusst support
When contacting support, please have the following information ready:
Tenant ID
Carrier name
Sample call SIP traces (if available)
Timestamps of failed calls (UTC)
Appendix A: Sample SDP
The following is an example SDP offer from the Trusst platform:
v=0
o=trusst 1234567890 1234567891 IN IP4 203.0.113.50
s=Trusst AI Voice Session
c=IN IP4 203.0.113.50
t=0 0
m=audio 16000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20SDP Field Reference
v=0
SDP version
o=
Origin/session identifier
s=
Session name
c=
Connection information (media destination IP)
t=
Timing (0 0 = permanent session)
m=
Media description (audio, port, protocol, payload types)
a=rtpmap
Codec mapping
a=fmtp
Format parameters (DTMF events 0-16)
a=ptime
Packetization time in milliseconds
Last updated