DEBIAN-SBC Mediant v2.0

Alarms & Notifications

Recent State Transitions
Active Sessions
concurrent
SRTP Encrypted:
Transcoding DSP
/ 48 slots
DSP Equiv. Load:
↑ UP Trunks (ISP)
Online
Carrier Gateway:
↓ DOWN Trunks (PBX)
Online
Internal LAN PBX:

Vertical SBC Topology Map (UP / DOWN)

Live Signaling & Media

Top: ↑ UP (Carriers / ISP)  |  Middle: Debian SBC Core  |  Bottom: ↓ DOWN (PBX / LAN)

↑ UP (ISP / Carrier SIP Trunks)
No UP Trunk (ISP) configured.
SIP Ingress Signalling & RTP Media
DEBIAN-SBC Core Appliance

B2BUA / Topology Hiding / Transcoder

SIP Proxy: 5060 (Active)
Media DSP: 2223 (Active)
Active Calls:
Transcode:
SIP Egress Signalling & Transcoded RTP
↓ DOWN (Internal PBX & LAN Trunks)
No DOWN Trunk (PBX) configured.

SBC Core Engines & Hardware Health

All Core Daemons Active
SIP Proxy Daemon
Kamailio 6.0
Listening: 0.0.0.0:5060 (UDP/TCP)
STATE: ACTIVE
Media Relay / Transcoder
RTPEngine 12.5
Control NG: 127.0.0.1:2223
STATE: ACTIVE
Appliance Board
Interface eth0 I/O

SIP Trunks (UP / DOWN Topology)

AudioCodes Topology Matrix

↑ UP = Uplink / ISP / Carrier Trunk  |  ↓ DOWN = Downlink / Internal PBX / LAN

Topology Trunk Type Trunk Name SIP Profile Egress Interface Gateways / Registrar Endpoint Status Actions
No SIP Trunks configured yet.
Add your first Trunk by selecting direction (UP = ISP or DOWN = PBX).

AudioCodes-grade SIP Trunk configuration supporting all trunk topologies: Peer-to-Peer, Carrier Registration, Inbound Registrar & Digest Authentication.

Unique alphanumeric label for this trunk.

AudioCodes Matrix: UP for Carriers, DOWN for Enterprise PBX.

Manage

Defines PRACK, Session Timers, and signaling normalization.

Automates Linux kernel static host routes (/32) for this trunk.

Manage Realms

Couples this trunk with dedicated RTP ports and network interface.

Periodic SIP OPTIONS pinging to monitor gateway liveness and auto-failover.

Standard P2P Peering Authentication

Standard Peer-to-Peer SIP trunks authenticate via IP Access Control / Firewall rules without SIP digest credentials.
If your provider or PBX challenges outgoing calls with 401 Unauthorized or 407 Proxy Authentication Required, switch the Trunk Topology above to "Digest Auth" or "Registration".

SIP Account Phone Number, DID, or Trunk Username.

SIP Digest authentication password.

Matches the realm in 401/407 challenge header.

Custom user in Contact header (defaults to Account DID).

All voice codecs, RTPEngine transcoding rules, SRTP encryption, and timers are governed by this SIP Profile.

Voice Codecs (Priority Order)
RTPEngine Transcoding
Transcode on codec mismatch Strict transcode to profile codecs Direct RTP pass-through
Media Encryption (SRTP)
Mandatory SDES / DTLS-SRTP Offer SRTP & Accept RTP Plain RTP (Unencrypted Audio)
DTMF Signaling Relay
Packet Time (ptime)
RTP Anchoring & Topology
•
Manage TLS Contexts

Automatically applies dedicated RSA/ECDSA Certificate, Root CA validation bundle, TLS version, and Ciphers when connecting via TLS.

All SBC Trunks & Real-Time Status

Complete topology matrix of all connections (UP Carriers & DOWN PBX) with live status

Topology Trunk Name SIP Profile Redundancy Endpoints (IP / FQDN:Port) Transport Status
No Trunks found matching criteria.

SIP Profiles & Signaling Definitions

AudioCodes IP Profile Engine

Codecs, Transcoding Policy, RTP Anchoring, DTMF Relay (RFC2833/INFO/Inband), SIP Extensions (PRACK / 100rel, Re-INVITE, UPDATE) and RFC 4028 Session Timers.

Trunk Binding Strategy
Each SIP Trunk is bound to a SIP Profile. Kamailio & RTPEngine automatically enforce these policies on each call leg.
Total Profiles:
Transcoding: DSP Dynamic
PRACK & RFC4028: Active
Profile Name & Description Allowed Codecs Media / Transcoding DTMF Relay SIP Extensions Session Timers (RFC4028) Assigned Trunks Actions
No SIP Profiles configured.
Add a new profile by clicking the "Add SIP Profile" button.

AudioCodes Profile

Configure Media/Transcoding policies, DTMF Relay, SIP Extensions (PRACK / UPDATE), and Session Timers (RFC 4028).

Select allowed codecs
Transcode Policy:

RTPEngine DSP transcoding policy when codecs between call legs differ.

RTP Anchoring Mode:

Media stream routing (Audio RTP) anchored through RTPEngine or direct peer-to-peer.

Media Encryption (SRTP):

RTP stream encryption via SRTP AES-GCM (mandatory for Microsoft Teams).

Packet Time (ptime) & Topology Hiding:

Audio packetization time (ptime ms) and internal network topology hiding.

DTMF Relay Mode:

Select DTMF signaling relay method towards this Trunk.

RFC 2833 Payload Type (Default 101):

Dynamic RTP payload type number (typical values: 101, 96, 102).

PRACK (RFC 3262 / 100rel Reliability)

Provisional responses reliability (18x Provisional Responses) and 100rel handling.

Re-INVITE Handling

In-dialog session modifications handling (Call Hold, Unhold, Call Transfer, Session Refresh).

SIP UPDATE (RFC 3311)

Early / Mid-dialog session parameters update before or after answer without re-INVITE.

Carriers typically require PRACK: supported and Re-INVITE: allowed for Music On Hold (MOH) and Call Transfer.
RFC 4028 SIP Session Timers Mode

Session-Expires & Min-SE headers negotiation for active call liveness detection.

Session-Expires & Min-SE (seconds):

Call refresh interval (Expires) and minimum allowable time (Min-SE).

Session Refresher Preference:

Role responsible for sending refresh requests (UAC, UAS or Auto).

Ringing Timeout (seconds):

Automatic call cancellation with 408 Request Timeout if unheeded.

Max Call Duration (seconds):

Maximum call duration to prevent orphan / ghost calls (e.g. 7200s = 2 hours).

codecs • • timers

Media Realms (RTP Per-Interface Management)

AudioCodes Mediant Multi-Homed RTP

Couple logical media realms with network interfaces, allocate dedicated UDP port ranges per interface, configure NAT public addresses, and enforce QoS.

Configured Realms

Coupled Interfaces

Allocated RTP Ports

UDP Ports

Audio Session Capacity

Concurrent

Couple network interface, assign dedicated RTP UDP port range, and set media parameters.

1. Realm Identification

Unique identifier referenced by Kamailio and RTPEngine for routing.

2. Network Interface Coupling (Πάντρεμα Interface)

Marries this media realm to a specific Linux physical, VLAN, or alias interface.

Bound Media IP (Auto-detected):

Advertised in outgoing SDP c= (Connection) and o= (Origin) lines when SBC sits behind 1:1 NAT (AWS EIP, carrier NAT).

3. RTP Port Allocation per Interface

Quick Standard Presets (Non-Overlapping Blocks):
Allocated UDP Sockets:
Concurrent Call Capacity:

4. Media Protocols, Latching & QoS

RTCP Multiplexing (RFC 5761 rtcp-mux): Multiplexes RTP and RTCP on the same single port pair.

Controls how RTPEngine learns and relays media for remote endpoints behind NAT.

Default 46 (Expedited Forwarding EF)

Default 34 (Assured Forwarding AF41)

TLS Contexts & Certificates

AudioCodes Mediant PKI

Manage Digital Certificates (Server / Client), Root CAs, Cipher Suites and TLS Profiles for SIP Trunks

Context Name & Description Min TLS Server / Client Certificate Root CA Bundle Verification / mTLS Ciphers Bound SIP Trunks Actions
No TLS Contexts configured.
Total Certificates:
Certificate Name & Type Common Name (CN) Issuer SANs (Alt Names) Key Size / Fingerprint Validity & Days Left In Use Actions
No installed digital certificates or Root CAs.
CSR Name Subject Common Name (CN) SANs (Alt Names) Key Size Generated Date Status Actions
No pending Certificate Signing Requests (CSR).

/etc/kamailio/tls.cfg Configuration File

Live production Kamailio TLS configuration file, synced from TLS Contexts and Certificates


      

AudioCodes Mediant PKI

Configure TLS encryption profile, create or attach certificates directly, Root CA and Cipher Suite policies.

Server / Client Certificate

Issue, generate or attach a digital certificate for this TLS Context

Subject CN:
Issuer:
Valid Until:

Trusted Root CA Bundle

Select an existing Root CA or import a new Root CA bundle

| |

Verifies peer cert is signed by Root CA.

Enforces mutual TLS authentication.

Presets:
Communicating with PKI / Let's Encrypt...

Automated Let's Encrypt Certificate (ACME v2)

ACME v2 Automated PKI

Issue automated SSL/TLS certificates with zero-downtime Kamailio SIP integration and daily auto-renewal.

HTTP-01 Standalone: Certbot binds port 80 to complete ACME token validation. Ensure public port 80 is forwarded to this SBC appliance.
HTTP-01 Webroot: Challenges are served via the appliance web server at /.well-known/acme-challenge/. Ideal when reverse proxies forward port 80.
DNS-01 Challenge: You will be given a DNS TXT record (_acme-challenge.<domain>) to add to your DNS zone. No inbound port 80 required!

Automatically generates a TLS Context named TLS_<Name> and reloads Kamailio.

Use testing environment without rate limits. (Uncheck for real production cert).

Action Required: Add DNS TXT Record

Please create the following TXT record at your DNS provider (e.g., Cloudflare, Route53, GoDaddy, BIND). Once added, wait 30 seconds for DNS propagation, then click "Verify & Complete".

Communicating with Let's Encrypt ACME...

Generate Self-Signed Certificate

Generate a new RSA Private Key and self-signed X.509 certificate for TLS/SIP.

Add all IP addresses and FQDN domain names that will be used by Trunks.

Generate Certificate Signing Request (CSR)

Generates RSA Private Key on the SBC and produces a CSR to submit to an external CA.

Import Certificate or Root CA Bundle

Import an existing Server/Client certificate or Trusted Root CA.

Required for Server or Client certificates to complete the TLS handshake.

Telephony Routing

Sequential routing engine organized into serial Groups and serial Rules with Drag & Drop or direct Number Order ID.

Sequential Group & Rule Execution Architecture

Groups execute in serial order (Group 1 → Group 2 → ...). Inside each group, rules are evaluated sequentially (Order 1, 2, ...). The first matching rule executes the Route To Trunk action. Reorder items with Drag & Drop (⠿) or pin an exact position with the Order ID input (existing rules automatically shift down by +1).

Assign routing group, set order ID, configure match attributes and target trunk.

Group & Sequential Execution Order

Assign rule to a serial routing execution group.

★ Auto-shift: If this Order ID exists, that rule and all below it shift down by +1.

Human-friendly route name.

Match Attributes

Originating SIP Trunk, or ANY (*) for all.

Matches SIP Request-URI Host / Domain ($rd). Use * for any host.

Dialed ($rU)
Quick:
Regex:
Caller ($fU)
Quick:
Regex:

SIP request method ($rm) to match.

Action

The destination SIP Trunk to forward and dispatch matched SIP messages.

Define serial routing group execution order and descriptive metadata.

Auto-shift: If occupied, existing groups shift down by +1.

Digit Manipulation

AudioCodes-style telephone number manipulation (Strip digits left/right, add prefix/suffix on Destination or Source patterns organized in sequential groups)

Assign manipulation group, set order ID, configure match rules and digit transformation.

Group & Sequential Execution Order

Assign rule to a serial manipulation execution group.

★ Auto-shift: If this Order ID exists, that rule and all below it shift down by +1.

Human-friendly rule description.

Match Rules

Ingress SIP Trunk from which call arrives.

Egress destination SIP Trunk determined by Telephony Routing.

Caller ($fU)
Quick:
Regex:
Dialed ($rU)
Quick:
Regex:

Originating IP address ($si) or Host. Use * for any host.

Target SIP Request-URI Host / Domain ($rd). Use * for any host.

Action (Digit Manipulation)

Number of digits to remove from start (left).

Number of digits to remove from end (right).

String / digits to prepend to the number.

String / digits to append to the number.

Live Interactive Transformation Preview Type any test number:
Transformed Result:

Define serial digit manipulation group execution order and descriptive metadata.

Auto-shift: If occupied, existing groups shift down by +1.

Header Manipulation

AudioCodes-style SIP message header normalization and manipulation (Inbound / Outbound serial groups assigned to SIP Trunks)

No header manipulation groups configured for .

Assign group, select RFC SIP Header & parameter, and configure transformation action.

Group & Sequential Execution Order

Assign rule to a serial manipulation group.

★ Auto-shift: If this Order ID exists, that rule and all below it shift down by +1.

Human-friendly rule description.

Target SIP Header & Parameter Component

Select any standard RFC SIP header or choose Custom.

Specify which component of the header to evaluate or modify.

Enter exact SIP header name.

Action (Header Transformation)
Condition
Quick:
Quick:

Define direction, trunk assignment, serial execution order, and metadata.

Auto-shift: If occupied, existing groups shift down by +1.

Select ALL Trunks or restrict this group to a specific SIP Trunk.

Interactive Call Route Simulator

Simulate any dialed number or call in real-time to inspect matched rules, required transcoding, and SIP header modifications!

Simulate Ingress SIP Call

SBC Routing & Transcoding Decision

Select call parameters and click "Test Routing & Media" to view call routing analysis.

Matched Route:
Destination IP & Port:
Destination Number ($rU):
Caller ID ($fU):
Applied Digit Manipulation Rules:
DSP Media Relay & Transcoding:
Applied SMM Rules (Header Rewrites):
Call Routing Decision: REJECTED (404 Not Found)

Evaluated Destination ($rU):
Calling Caller ID ($fU):
ROUTING TROUBLESHOOTING:

• Check if a rule exists in Telephony Routing matching pattern .

• Ensure the rule is Enabled and allows the selected Ingress Trunk.

• Check if caller ID pattern restriction matches (or is *).

Network Interfaces

Network interfaces management

Network Interfaces List

Interface Type Status IPv4 Address & Subnet Gateway MAC / MTU Role / Trunk Binding Traffic (RX / TX) Actions

Add Network Interface / VLAN

Create an 802.1Q VLAN (e.g. Carrier Voice VLAN 835/100) or an IP Alias (secondary IP on same NIC).

Configure or change primary IP directly on physical parent interface ().

Parent () IP:
Target Interface Name:

Configure Interface:

Edit IP address, netmask, gateway and DNS. Applied live to the kernel and to /etc/network/interfaces.

Changes are applied immediately to network interfaces without disconnecting current sessions and saved to /etc/network/interfaces.

Routing Table Management

Linux kernel routing table and automated Trunk static routes

Routing Table Management

Includes default gateways, connected subnets, and automated host routes (/32) generated by each Trunk.

Destination Gateway Interface Routing Type Actions

Add Static Route

Add a kernel static route via a Gateway IP, an Output Interface, or both.

Specify destination CIDR prefix (e.g. 192.168.100.0/24) or single host IPv4.

Next-hop router IP address.

Direct egress network interface.

You can specify a Gateway IP, an Interface, or both together.

VoIP Diagnostics, Live PCAP & Call Ladder

AudioCodes-grade diagnostic tools: Live Packet Capture (Wireshark) and SIP Call Flow Diagrams

Live Wireshark Packet Capture (SIP & RTP)

Real-time packet capture for Wireshark traffic inspection

Recording to:
Size:

Saved PCAP Captures:

No saved pcap capture files yet.

Interactive SIP Call Flow (Ladder Diagram)

Generated Kamailio Configuration

Auto-generated `/root/sbc-appliance/config/kamailio_generated.cfg`


    

SIP Syslog Analyzer

AudioCodes Mediant Edition LIVE WS STREAM OFFLINE

Real-time SIP packet capture, Call-ID correlation ladder diagrams, and live Kamailio syslog monitoring.

Total Calls
0
Connected
0
In Progress
0
Failed / Rejected
0
Syslog Messages
0
Call Sessions
Correlated by Session ID
Call Flow Sequence Diagram (Ladder)
Trunk 1
SBC
Trunk 2
SIP Message Inspector
Live Syslog Stream (/var/log/kamailio.log)

User & Access Management

Manage local operator accounts and configure Active Directory / LDAP corporate directory integration.

Local database accounts stored securely with PBKDF2-SHA256 password hashing.
User Full Name / Email Role Status Auth Source Last Login Actions

Active Directory & LDAP Corporate Authentication

Authenticate operators directly against Microsoft Active Directory, OpenLDAP, or FreeIPA with RBAC group mapping.

Quick Schema Presets:

1. Server & Base Directory

Specify protocol (ldap:// or ldaps://), hostname or IP, and port (389 for plain/STARTTLS, 636 for LDAPS).

Root Distinguished Name for all directory searches.

2. Service Account (Bind DN)

Account used to query user entries and group memberships. Leave blank for Anonymous bind.

Service account password. Masked for security.

3. User Search Query

Use {username} as the placeholder for the input login name.

4. Role & Group Mapping (RBAC)

Comma-separated list of group names or DNs granted the admin role.

5. Access Whitelist & Fallback Rules

If set, users MUST belong to at least one of these groups to be granted login access.

Restrict login to specific directory usernames.

SBC Fault & Alarm Management Center

Real-time status monitoring and chronological audit logging of all SBC events, link drops, and recoveries.

Active Alarms
Critical Alerts
Major / Warning
History Log Count
Severity Source & Entity Alarm Title / Description Raised At Duration Status Action
Timestamp Event Severity Source Entity Event Summary & Details
Automatic polling every 3 seconds • SIP OPTIONS & Link carrier state detection

DEBIAN-SBC Mediant

Appliance Sign In • Local & Active Directory / LDAP

Default: admin / admin

Add Local User Account

Edit User:

Change Password:

Test LDAP / Active Directory Connection

Verify network socket reachability, service credentials bind, and optionally simulate user query & group-to-role mapping.

Reachability
Service Bind
User Query
Mapped Role