SIP Profiles Module Documentation

Navigation & Access
Screenshots & Visual Interface
Module Overview (Technical)
Module Overview (Commercial/Business)
Module Overview (End User/Administrator)
User Roles & Key Capabilities
Configuration Categories
Common Settings Reference
Common Scenarios & Examples
Limitations & Important Notes
Troubleshooting Tips
Glossary
Model Context Protocol (MCP) AI Integration

To access the SIP Profiles (SIP Settings) configuration module:

Log in to the Ring2All Web Portal (https://<domain-or-ip>/login).
In the left navigation sidebar, expand Settings.
Under Technology, click SIP Settings (/settings/technology/sip).
To modify an existing profile (such as internal, external, or webrtc), click on the profile row or the Edit action button (/settings/technology/sip/:id).

Screenshots & Visual Interface

SIP Profiles Directory & Status Overview

Centralized inventory of Telephony Server Sofia SIP profiles, showing profile names, bindings, categories, active RTP media IP bindings, SIP ports, and operational runtime status. SIP Profiles List

SIP Profile Configuration & Parameter Management

Full-featured configuration interface providing granular control over Sofia SIP stack parameters including SIP bind IP, port, TLS/WSS encryption, NAT traversal, codec negotiation, ACL filters, and timer parameters. SIP Profile Form

1. Module Overview (Technical)

What Are SIP Profiles?

SIP Profiles is a Telephony Server configuration module that manages SIP profile settings. SIP profiles define how Telephony Server handles SIP signaling, including transport protocols, codecs, NAT handling, security, and registration behavior.

Architecture

┌─────────────────────────────────────────────────────────────────┐
│                  SIP Profiles Architecture                      │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  Admin Panel                                                    │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │        SIP Profiles Page                                 │   │
│  │                                                          │   │
│  │  Profiles: [internal] [external] [custom]                │   │
│  │                                                          │   │
│  │  Categories:                                             │   │
│  │  ├─ General Settings                                     │   │
│  │  ├─ Transport (SIP/TLS)                                  │   │
│  │  ├─ Media & Codecs                                       │   │
│  │  ├─ NAT / ACL                                            │   │
│  │  ├─ Security & Auth                                      │   │
│  │  ├─ Registration                                         │   │
│  │  ├─ Call Handling                                        │   │
│  │  └─ Advanced                                             │   │
│  │                                                          │   │
│  └──────────────────────────────────────────────────────────┘   │
│       │                                                         │
│       ▼ Configuration saved                                     │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │        public.sip_profiles                               │   │
│  │        public.sip_profile_settings                       │   │
│  │                                                          │   │
│  └──────────────────────────────────────────────────────────┘   │
│       │                                                         │
│       ▼ Telephony Server XML generated                               │
│  ┌──────────────────────────────────────────────────────────┐   │
│  │        Telephony Server mod_sofia                              │   │
│  │                                                          │   │
│  │  <profile name="internal">                               │   │
│  │    <param name="sip-port" value="5060"/>                │   │
│  │    <param name="rtp-timeout-sec" value="300"/>          │   │
│  │    ...                                                   │   │
│  │  </profile>                                              │   │
│  │                                                          │   │
│  └──────────────────────────────────────────────────────────┘   │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

2. Module Overview (Commercial/Business)

Business Value

SIP Profiles provides centralized SIP configuration:

Without SIP Profiles	With SIP Profiles
Manual XML editing	Web interface
CLI configuration	Category-based UI
Error-prone edits	Validated settings
Server access needed	Browser-based

Use Cases

Transport Configuration
- Configure SIP ports
- Enable TLS encryption
NAT Handling
- Configure STUN/TURN
- Set external IP
Codec Management
- Configure codec preferences
- Enable/disable codecs
Security Setup
- Configure authentication
- Set registration policies

Feature Highlights

Feature	Benefit
Web Interface	No CLI needed
Categories	Organized settings
Default Values	Easy reset
Multiple Profiles	Internal/External
Search	Find settings fast
Validation	Prevent errors

3. Module Overview (End User/Administrator)

What Can You Do?

Create and manage SIP profiles
Configure SIP transport settings
Set media and codec preferences
Configure NAT traversal
Set security parameters
Manage registration behavior
Configure call handling options

SIP Profiles Interface

┌─────────────────────────────────────────────────────────────────┐
│              SIP Profiles                                       │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  [+ Create SIP Profile]                                         │
│                                                                 │
│  ┌───────────────────────────────────────────────────────────┐  │
│  │ Profile     │ Description       │ Settings │ Status      │  │
│  ├─────────────┼───────────────────┼──────────┼─────────────┤  │
│  │ internal    │ Internal SIP      │ 45       │ 🟢 Enabled  │  │
│  │ external    │ External/Trunks   │ 52       │ 🟢 Enabled  │  │
│  │ secure      │ TLS-only profile  │ 48       │ 🟢 Enabled  │  │
│  └───────────────────────────────────────────────────────────┘  │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

Profile Edit View

┌─────────────────────────────────────────────────────────────────┐
│              Edit SIP Profile: internal                         │
├─────────────────────────────────────────────────────────────────┤
│                                                                 │
│  Profile Name: [internal                         ]              │
│  Description: [Internal SIP for extensions       ]              │
│  Enabled: ✓                                                     │
│                                                                 │
│  🔍 [Search settings...                               ]        │
│                                                                 │
│  ▼ General Settings (12 parameters)                             │
│  ├─ context: [default           ] [↺ Reset]                    │
│  ├─ dialplan: [XML              ] [↺ Reset]                    │
│  └─ user-agent: [Ring2All       ] [↺ Reset]                    │
│                                                                 │
│  ▼ Transport (SIP/TLS) (15 parameters)                          │
│  ├─ sip-port: [5060             ] [↺ Reset]                    │
│  ├─ sip-ip: [auto               ] [↺ Reset]                    │
│  ├─ tls: [false                 ] [↺ Reset]                    │
│  └─ tls-cert-dir: [/etc/certs   ] [↺ Reset]                    │
│                                                                 │
│  ▶ Media & Codecs (8 parameters)                                │
│  ▶ NAT / ACL (10 parameters)                                    │
│  ▶ Security & Auth (7 parameters)                               │
│  ▶ Registration (6 parameters)                                  │
│  ▶ Call Handling (5 parameters)                                 │
│  ▶ Advanced (12 parameters)                                     │
│                                                                 │
│  [Save] [Cancel]                                                │
│                                                                 │
└─────────────────────────────────────────────────────────────────┘

Quick Tips

[!TIP] Reset to Default: Click ↺ to reset a parameter to its default value.

[!TIP] Search: Use search to find specific settings across categories.

[!CAUTION] Restart Required: Profile changes require Telephony Server reload.

🎯 User Roles & Key Capabilities

Role	Permissions	Key Capabilities
Super Administrator	Full Access (`read`, `write`, `delete`, `restart`)	Manage Sofia SIP profiles (`internal`, `external`), modify network bind addresses, configure TLS certificates, and restart Sofia profiles.
PBX / VoIP Engineer	Operational Management (`read`, `write`)	Configure SIP timers, adjust NAT IP rewrites, modify codec lists (OPUS, PCMU, G729), and inspect live profile registration state.
Support Specialist	Read & Diagnostic (`read`, `status`)	View active bind ports, inspect Sofia status metrics, and execute live SIP trace diagnostics without mutating XML configurations.
Tenant Administrator	Restricted Read	View assigned technology profile settings in a multi-tenant PBX partition.

4. Configuration Categories

General Settings

Purpose	Common Parameters
Basic profile behavior	context, dialplan, user-agent

Transport (SIP/TLS)

Purpose	Common Parameters
SIP signaling layer	sip-port, sip-ip, tls, tls-cert-dir

Media & Codecs

Purpose	Common Parameters
Audio/video settings	codec-prefs, rtp-timer-name

NAT / ACL

Purpose	Common Parameters
NAT traversal	ext-rtp-ip, ext-sip-ip, apply-inbound-acl

Security & Auth

Purpose	Common Parameters
Authentication	challenge-realm, auth-calls

Registration

Purpose	Common Parameters
Device registration	accept-blind-reg, disable-register

Call Handling

Purpose	Common Parameters
Call behavior	rtp-timeout-sec, hold-music

Advanced

Purpose	Common Parameters
Expert settings	debug, sip-trace, nonce-ttl

5. Common Settings Reference

Essential Parameters

Parameter	Default	Description
context	default	Dialplan context
sip-port	5060	SIP UDP/TCP port
sip-ip	auto	Listen IP address
rtp-ip	auto	RTP media IP
user-agent	Telephony Server	SIP user agent

TLS Settings

Parameter	Default	Description
tls	false	Enable TLS
tls-cert-dir		Certificate directory
tls-version	tlsv1.2	TLS version
tls-bind-params		TLS binding options
ws-binding	127.0.0.1:5066	WebSocket (plain WS) — localhost only for Nginx proxy
wss-binding	(disabled)	WSS — disabled, TLS handled by Nginx on port 443

NAT Settings

Parameter	Default	Description
ext-rtp-ip		External RTP IP
ext-sip-ip		External SIP IP
local-network-acl		Local network ACL
NDLB-force-rport		Force rport

Timeout Settings

Parameter	Default	Description
rtp-timeout-sec	300	RTP inactivity timeout
rtp-hold-timeout-sec	1800	Hold timeout
session-timeout	1800	Session timeout

6. Common Scenarios & Examples

Scenario 1: Enable TLS

Edit SIP profile
Expand “Transport (SIP/TLS)”
Set tls = true
Configure tls-cert-dir
Save and reload

Scenario 2: Configure NAT for Cloud

Edit profile
Expand “NAT / ACL”
Set ext-rtp-ip = public IP
Set ext-sip-ip = public IP
Save and reload

Scenario 3: Change SIP Port

Edit profile
Expand “Transport”
Change sip-port
Save and reload

Scenario 4: Enable Debug

Edit profile
Expand “Advanced”
Set debug = true
Save and reload
Check logs

Scenario 5: WebRTC Configuration

WebRTC uses Nginx as a TLS proxy to Telephony Server:

Browser → wss://domain/ws (Nginx, port 443)
       → ws://127.0.0.1:5066 (Telephony Server, plain WS)

Edit internal profile
Expand “Transport (SIP/TLS)”
Set ws-binding = 127.0.0.1:5066 (localhost only)
Disable wss-binding (Nginx handles TLS termination)
Save and reload

[!NOTE] No TLS certificates are needed on Telephony Server for WebRTC. Nginx handles all WSS connections on port 443 and forwards them as plain WebSocket to Telephony Server on localhost.

7. Limitations & Important Notes

Technical Notes

[!NOTE] Profile Restart: Changes require sofia profile <name> restart.

[!NOTE] Two Profiles: Typically “internal” and “external” profiles.

[!WARNING] TLS Certificates: Must be valid and accessible.

Best Practices

Backup First: Export profile before major changes
Test Changes: Use debug mode
Document Changes: Track what you modified
Monitor After: Watch for registration issues
Use Defaults: Only change what you need

Common Profiles

Profile	Purpose
internal	Extensions, internal devices
external	Gateways, trunks
secure	TLS-only connections

8. Troubleshooting Tips

Common Issues

Symptom	Possible Cause	Solution
No registrations	Wrong SIP port	Check sip-port setting
One-way audio	NAT issues	Configure ext-rtp-ip
TLS fails	Bad certificates	Check tls-cert-dir
Timeouts	Short timeout values	Increase rtp-timeout-sec
Auth failures	Wrong challenge-realm	Check security settings

Diagnostic Commands

Telephony Server CLI:

# Show profile status
fs_cli -x "sofia status profile internal"

# View profile settings
fs_cli -x "sofia profile internal gwlist"

# Restart profile
fs_cli -x "sofia profile internal restart reloadxml"

# Enable SIP trace
fs_cli -x "sofia profile internal siptrace on"

Check Profile Settings

SELECT
  sp.name,
  sps.setting_name,
  sps.setting_value
FROM public.sip_profiles sp
JOIN public.sip_profile_settings sps ON sp.id = sps.sip_profile_id
WHERE sp.name = 'internal';

9. Glossary

Term	Definition
SIP Profile	Telephony Server SIP endpoint configuration
mod_sofia	Telephony Server SIP module
TLS	Transport Layer Security
NAT	Network Address Translation
RTP	Real-time Transport Protocol
ACL	Access Control List
ext-rtp-ip	External RTP IP for NAT

Model Context Protocol (MCP) AI Integration

The Ring2All PBX platform integrates deep AI assistance via the Model Context Protocol (MCP). The Sofia SIP technology stack exposes intelligent tools that permit the PBX AI Copilot to safely query profile parameters, inspect real-time SIP engine socket status, and diagnose registration or port binding conflicts.

Available MCP Tools

Tool Name	Operation Type	RBAC Risk Level	Description
`list_sip_profiles`	Read / Query	`low`	Lists all base Sofia SIP profiles (e.g. `internal`, `external`) with their IP bindings, SIP ports, TLS bindings, and status.
`get_sip_profile_status`	Live Engine Diagnostic	`low`	Retrieves real-time runtime status, SIP URIs, registered endpoints count, and socket metrics directly from Telephony Server CLI (`sofia status profile <name>`).

Tool Input Schemas & Parameters

1. `list_sip_profiles`

{
  "name": "list_sip_profiles",
  "description": "List all base Sofia SIP profiles (internal, external, etc.) running on the PBX core engine with bind ports and TLS settings.",
  "inputSchema": {
    "type": "object",
    "properties": {
      "search": {
        "type": "string",
        "description": "Filter profile by name or description (e.g. 'internal', 'external')."
      }
    }
  }
}

2. `get_sip_profile_status`

{
  "name": "get_sip_profile_status",
  "description": "Get live runtime status and registration metrics of a Sofia SIP Profile directly from Telephony Server CLI.",
  "inputSchema": {
    "type": "object",
    "properties": {
      "profile": {
        "type": "string",
        "description": "Sofia SIP profile name (e.g. 'internal', 'external')."
      }
    },
    "required": ["profile"]
  }
}

Natural Language Prompts

User Request	Invoked MCP Tool	Expected AI Response
“Show me all active Sofia SIP profiles and their bind ports.”	`list_sip_profiles`	Returns table of profiles (`internal` on 5060, `external` on 5080) with TLS ports and operational states.
“Check if the internal SIP profile is currently running in Telephony Server.”	`get_sip_profile_status({ profile: "internal" })`	Reports live Telephony Server CLI status, active registrations count, and RTP binding IP.
“Are there any errors on the external SIP trunk profile?”	`get_sip_profile_status({ profile: "external" })`	Details profile status, external NAT IP, and detects if the socket is bound or in error.

Multi-Tenant & Security Safeguards

Strict Domain Isolation: Sofia base profiles operate at the core engine level; access is restricted to authorized administrative tenants and roles.
Sanitized Parameter Execution: Profile names are strictly filtered ([a-zA-Z0-9_\-]) before dispatching commands to Telephony Event Socket (ESL), preventing command injection.
Read-Only Diagnostics: Status tools perform diagnostic queries only; destructive profile restarts require elevated Super Admin permissions with explicit confirmation.
Audit Logging: Every AI query into SIP profile telemetry is logged with user ID, tenant ID, and timestamp.

Documentation last updated: January 2026