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VOS3000 Session Timer: Complete Easy Guide to SIP Keep-Alive Configuration

VOS3000 Session Timer: Complete Guide to SIP Keep-Alive Configuration

VOS3000 session timer is a critical mechanism for maintaining call stability and preventing “zombie calls” that consume system resources. Based on RFC 4028 specifications, the session timer functionality in VOS3000 2.1.9.07 ensures that active VoIP sessions are properly monitored while failed or hung calls are detected and cleaned up automatically. This comprehensive guide covers all session timer parameters, NAT keep-alive configuration, and troubleshooting procedures based on the official VOS3000 manual.

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🔍 What is VOS3000 Session Timer?

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.3 (Page 213)

The VOS3000 session timer implements the SIP Session Timer mechanism defined in RFC 4028. This protocol extension addresses a fundamental problem in SIP-based VoIP systems: the inability to detect when a call has failed at one endpoint while the other endpoint believes the call is still active. These “zombie calls” can persist indefinitely, consuming system resources, occupying call capacity, and causing billing discrepancies.

📊 The Zombie Call Problem

🚨 Scenario❌ Without Session Timer✅ With Session Timer
Endpoint Power FailureCall remains “active” indefinitely in systemSession expires, call terminated cleanly
Network DisconnectionNo notification, resources wastedRefresh fails, session cleaned up
Device CrashZombie call persists for hours/daysMaximum session duration enforced
NAT TimeoutOne-way audio, confused stateSession refresh detects failure
Billing ImpactIncorrect CDR duration, revenue lossAccurate call termination timing

⚙️ VOS3000 Session Timer Parameters Complete Reference

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Pages 229-239)

VOS3000 provides a comprehensive set of session timer parameters that control how the softswitch monitors and maintains active SIP sessions. These parameters are configured in the System Parameters section and affect all SIP-based communications.

📊 Core Session Timer Parameters Table

⚙️ Parameter📊 Default📏 Range📝 Description📖 Manual Page
SS_SIP_SESSION_TTL60060-86400 secDetecting SIP connected status interval (Session-Expires value)230
SS_SIP_SESSION_UPDATE_SEGMENT22-10Divisor for refresh interval calculation (TTL/segment)230
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP00-3600 secTerminate session before actual timeout (margin)230
SS_SIP_NO_TIMER_REINVITE_INTERVAL72000-86400 secMaximum call duration for non-timer endpoints230
SS_SIP_SESSION_MIN_SE9090-3600 secMinimum session expires value per RFC 4028231

📊 Session Timer Refresh Calculation

📐 Session Timer Refresh Interval Formula

Refresh Interval = SS_SIP_SESSION_TTL ÷ SS_SIP_SESSION_UPDATE_SEGMENT

Example with Defaults:600 ÷ 2 = 300 seconds (5 minutes)
First Refresh Attempt:At 5 minutes into the call
Session Expires If:No response to refresh within TTL period

📡 NAT Keep-Alive Configuration Deep Dive

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.2 (Pages 212-213)

NAT (Network Address Translation) devices maintain binding tables that map internal private IP addresses to external public addresses. These bindings have a timeout period, typically ranging from 30 to 300 seconds depending on the device. When a binding expires without traffic, incoming calls cannot reach the endpoint behind NAT.

📊 NAT Keep-Alive Parameters Table

⚙️ Parameter📊 Default📏 Range📝 Function📖 Page
SS_SIP_NAT_KEEP_ALIVE_MESSAGEHELLOText stringContent of NAT keep-alive UDP packet212
SS_SIP_NAT_KEEP_ALIVE_PERIOD3010-86400 secInterval between keep-alive transmissions212
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL5001-10000 msDelay between individual keep-alive packets in batch212
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME30001-10000Number of keep-alive packets sent per batch cycle212

🔄 How NAT Keep-Alive Works in VOS3000

VOS3000 NAT Keep-Alive Operation Flow:
=======================================

SCENARIO: Endpoint behind NAT firewall
┌─────────────────────────────────────────────────────────────────────────────┐
│                                                                             │
│  ENDPOINT                    NAT DEVICE                   VOS3000 SERVER    │
│  (192.168.1.100)            (Public IP)                  (Softswitch)       │
│                                                                             │
│  1. REGISTER ───────────────────────────────────────────────────────────►  │
│     (Via: 192.168.1.100)                                                    │
│                                                                             │
│  2. VOS3000 Records:                                                         │
│     - Received IP: Public NAT IP                                            │
│     - Received Port: NAT mapped port                                        │
│     - Contact: Internal IP (via Contact header)                             │
│                                                                             │
│  3. NAT BINDING TABLE:                                                       │
│     Internal: 192.168.1.100:5060 → External: PublicIP:45678                │
│                                                                             │
│  4. KEEP-ALIVE MESSAGE (every 30 seconds):                                  │
│     ◄─────────────────────────────────────────────────────────────────────  │
│     UDP packet "HELLO" to PublicIP:45678                                    │
│                                                                             │
│  5. NAT BINDING REFRESHED:                                                   │
│     - Timer resets to 30+ seconds                                           │
│     - Binding remains active                                                │
│                                                                             │
│  6. INCOMING CALL:                                                           │
│     ◄─────────────────────────────────────────────────────────────────────  │
│     INVITE reaches endpoint successfully!                                   │
│                                                                             │
└─────────────────────────────────────────────────────────────────────────────┘

IMPORTANT: If SS_SIP_NAT_KEEP_ALIVE_MESSAGE is empty, keep-alive is DISABLED!

🔧 VOS3000 Session Timer Configuration Guide

📍 Navigation to System Parameters

StepNavigation PathAction
1Operation managementClick main menu
2Softswitch managementSelect softswitch node
3Additional settingsRight-click → Additional settings
4System parameter tabFind session timer parameters
5Modify valuesEdit desired parameters
6Apply changesClick OK to save
🏢 Scenario⏱️ SESSION_TTL🔄 SEGMENT🚫 NO_TIMER_INTERVAL📡 NAT_PERIOD
Standard VoIP Wholesale600 (10 min)20 (disabled)30 sec
Call Center Operations900 (15 min)314400 (4 hrs)20 sec
Mobile/Unstable Networks300 (5 min)23600 (1 hr)15 sec
Enterprise PBX1200 (20 min)228800 (8 hrs)30 sec
High-Security Environment180 (3 min)21800 (30 min)10 sec

📊 Session Timer Message Flow Diagram

VOS3000 Session Timer - Complete Call Flow with Refresh:
=========================================================

CALLER                          VOS3000                         CALLEE
  │                               │                               │
  │  1. INVITE                    │                               │
  │  Session-Expires: 600         │                               │
  │  Min-SE: 90                   │                               │
  │──────────────────────────────►│                               │
  │                               │  2. INVITE (forwarded)        │
  │                               │  Session-Expires: 600         │
  │                               │──────────────────────────────►│
  │                               │                               │
  │                               │  3. 200 OK                    │
  │                               │  Session-Expires: 600         │
  │                               │◄──────────────────────────────│
  │  4. 200 OK                    │                               │
  │  Session-Expires: 600         │                               │
  │◄──────────────────────────────│                               │
  │                               │                               │
  │  5. ACK                       │                               │
  │──────────────────────────────►│  6. ACK                       │
  │                               │──────────────────────────────►│
  │                               │                               │
  │           ═════════════════════════════════════════           │
  │           ║    CALL ACTIVE - AUDIO FLOWING           ║        │
  │           ═════════════════════════════════════════           │
  │                               │                               │
  │  [5 minutes into call]        │                               │
  │                               │                               │
  │  7. UPDATE (session refresh)  │                               │
  │  Session-Expires: 600         │                               │
  │◄──────────────────────────────│                               │
  │  8. 200 OK                    │                               │
  │  Session-Expires: 600         │                               │
  │──────────────────────────────►│                               │
  │                               │  9. UPDATE (session refresh)  │
  │                               │──────────────────────────────►│
  │                               │  10. 200 OK                   │
  │                               │◄──────────────────────────────│
  │                               │                               │
  │           ═════════════════════════════════════════           │
  │           ║    SESSION REFRESHED SUCCESSFULLY       ║        │
  │           ═════════════════════════════════════════           │
  │                               │                               │
  │  [If refresh fails]           │                               │
  │                               │                               │
  │  11. BYE (session timeout)    │                               │
  │◄──────────────────────────────│  12. BYE (session timeout)    │
  │                               │──────────────────────────────►│
  │                               │                               │
  │  CDR: Termination Reason = "Session Timeout"                 │
  │                               │                               │

🚨 Session Timer Troubleshooting Guide

📊 Common Problems and Solutions

🚨 Symptom🔍 Root Cause✅ Solution📖 Reference
Calls drop at exactly 30 secondsNAT binding timeout, not session timerEnable NAT keep-alive, reduce period to 15-20sPage 212
Calls drop at 5-minute intervalsSession refresh failingCheck if endpoint supports re-INVITE/UPDATEPage 213
“422 Session Interval Too Small” errorSession-Expires below minimumIncrease SS_SIP_SESSION_MIN_SE or TTLPage 231
No incoming calls after idle periodNAT binding expiredVerify NAT keep-alive is enabled and workingPage 212
Re-INVITE rejected with 491Glare condition (simultaneous re-INVITEs)Normal – VOS3000 will retry automaticallyPage 213
Zombie calls still occurringSession timer not negotiatedCheck NO_TIMER_REINVITE_INTERVAL settingPage 230

🔧 Debug Trace Analysis for Session Timer

VOS3000 Debug Trace - Session Timer Analysis:
==============================================

Step 1: Enable Debug Trace
Navigation: System → Debug trace
Enable: Check "On"
Set duration: 10-30 minutes

Step 2: Look for Session Timer Headers in SIP Messages:
───────────────────────────────────────────────────────

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK12345
From: ;tag=abc123
To: 
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: 
Session-Expires: 600;refresher=uac    ← SESSION TIMER HEADER
Min-SE: 90                            ← MINIMUM SESSION EXPIRES
Content-Type: application/sdp
Content-Length: ...

Step 3: Check 200 OK Response:
──────────────────────────────
SIP/2.0 200 OK
...
Session-Expires: 600;refresher=uac    ← CONFIRMED SESSION TIMER
...

Step 4: Look for Session Refresh Messages (UPDATE or re-INVITE):
────────────────────────────────────────────────────────────────

UPDATE sip:[email protected]:5060 SIP/2.0
...
Session-Expires: 600                    ← REFRESHING SESSION
...

Step 5: If No Session Timer Headers Found:
──────────────────────────────────────────
- Endpoint does not support RFC 4028
- VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL
- Maximum call duration will be enforced

📊 Session Timer vs NAT Keep-Alive Comparison

📊 Aspect⏱️ Session Timer📡 NAT Keep-Alive
Primary PurposeDetect failed calls, prevent zombie sessionsMaintain NAT bindings for incoming calls
RFC StandardRFC 4028 (SIP Session Timer)NAT traversal best practices
Protocol UsedSIP re-INVITE or UPDATE messagesUDP packets or SIP messages
When ActiveDuring active call (after 200 OK)While endpoint is registered
DirectionBidirectional (negotiated refresh)Server to endpoint (unidirectional)
Default Interval600 seconds (10 minutes)30 seconds
Failure ResultCall terminated, CDR updatedIncoming calls may fail
Endpoint Support RequiredYes (RFC 4028 compliance)No (transparent to endpoint)

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❓ Frequently Asked Questions about VOS3000 Session Timer

What happens if an endpoint doesn’t support session timer?

VOS3000 will use the SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter to limit the maximum call duration. This ensures that zombie calls cannot persist indefinitely even when the endpoint doesn’t support RFC 4028. Set this value based on your business requirements (default is 7200 seconds or 2 hours).

Why are my calls dropping exactly at 30 seconds?

30-second call drops are almost always caused by NAT binding timeout, not session timer issues. The solution is to enable NAT keep-alive by setting SS_SIP_NAT_KEEP_ALIVE_MESSAGE to a value like “HELLO” and reducing SS_SIP_NAT_KEEP_ALIVE_PERIOD to 15-20 seconds. Also check if SIP ALG is enabled on your router (it should be disabled).

What is the difference between re-INVITE and UPDATE for session refresh?

Both methods can be used for session refresh. UPDATE is generally preferred because it doesn’t modify the SDP session parameters, while re-INVITE also renegotiates media. VOS3000 automatically selects the appropriate method based on endpoint capabilities and configuration.

How do I calculate the optimal session timer refresh interval?

The refresh interval equals SS_SIP_SESSION_TTL divided by SS_SIP_SESSION_UPDATE_SEGMENT. With defaults (600 ÷ 2 = 300 seconds), VOS3000 sends a refresh every 5 minutes. For mobile networks, consider 300 ÷ 2 = 150 seconds for faster failure detection.

Can session timer prevent billing fraud?

Session timer helps prevent zombie calls that could result in incorrect CDR durations, but it’s not a fraud prevention mechanism. For fraud protection, implement proper account limits, IP restrictions, and monitor for unusual calling patterns using VOS3000’s built-in reports.

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VOS3000 SIP Session Timer: Complete Keep-Alive & Session Management Important Guide

VOS3000 SIP Session Timer: Complete Keep-Alive & Session Management Guide

VOS3000 SIP session timer is essential for maintaining reliable VoIP calls and preventing “zombie calls” that waste resources. By implementing RFC 4028 session timers and NAT keep-alive mechanisms, VOS3000 ensures that active calls are properly monitored and terminated calls are detected quickly. This comprehensive guide covers all session timer and keep-alive features based on official VOS3000 2.1.9.07 documentation.

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🔍 Understanding VOS3000 SIP Session Timer

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.3 (Page 213)

The SIP Session Timer, defined in RFC 4028, provides a mechanism to detect failed calls that would otherwise remain “hung” in the system. Without session timers, calls that lose one-way audio or have endpoint failures may continue to exist in the system, consuming resources and potentially causing billing errors.

📊 Why Session Timers Matter

ProblemWithout Session TimerWith Session Timer
Zombie CallsCalls remain active indefinitely after endpoint failureFailed endpoints detected, calls cleaned up
Resource WasteSystem resources consumed by dead sessionsResources freed when session expires
Billing ErrorsIncorrect long-duration billing for dead callsAccurate call termination timing
NAT IssuesNAT bindings expire causing call dropsKeep-alive maintains NAT bindings

⚙️ VOS3000 SIP Session Timer Parameters

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Page 230-231)

📊 Core Session Timer Parameters

ParameterDefaultRangeDescription
SS_SIP_SESSION_TTL600secondsDetecting SIP connected status interval
SS_SIP_SESSION_UPDATE_SEGMENT22-10SIP timer re-INVITE/UPDATE interval segment
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP0secondsSession timer early hangup before timeout
SS_SIP_NO_TIMER_REINVITE_INTERVAL7200secondsMax conversation time for non-timer SIP caller

📐 How Session Timer Works (VOS3000 SIP Session Timer)

VOS3000 SIP Session Timer Operation:
================================

1. Call Establishment:
- INVITE with Session-Expires header (if supported)
- VOS3000 records session timer requirements

2. Session Refresh:
- Re-INVITE or UPDATE sent at regular intervals
- Interval = SS_SIP_SESSION_TTL / SS_SIP_SESSION_UPDATE_SEGMENT
- Default: 600 / 2 = 300 seconds (5 minutes)

3. Session Monitoring:
- If refresh fails, session is considered dead
- Call is terminated after timeout
- CDR updated with proper end reason

4. Non-Timer Endpoints:
- For SIP endpoints without timer support
- VOS3000 uses SS_SIP_NO_TIMER_REINVITE_INTERVAL
- Default 7200 seconds (2 hours) maximum call duration

Example Flow with SS_SIP_SESSION_TTL = 600:
===========================================
Time 0:00 - Call established
Time 5:00 - Re-INVITE/UPDATE sent (refresh attempt)
Time 5:01 - 200 OK received (refresh successful)
Time 10:00 - Re-INVITE/UPDATE sent
Time 10:01 - 200 OK received
...continues for duration of call

If refresh fails:
Time 10:00 - Re-INVITE/UPDATE sent
Time 10:30 - No response (timeout)
Time 10:30 - Call terminated
Time 10:30 - CDR records "Session timeout"

📡 NAT Keep-Alive Configuration

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.2 (Page 212-213)

NAT keep-alive ensures that NAT bindings remain active for devices behind NAT devices. Without proper keep-alive, incoming calls may fail because the NAT mapping has expired.

⚙️ NAT Keep-Alive Parameters

ParameterDefaultRangeDescription
SS_SIP_NAT_KEEP_ALIVE_MESSAGEHELLOtextContent of NAT keep-alive message
SS_SIP_NAT_KEEP_ALIVE_PERIOD3010-86400 secNAT keep-alive message sending period
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL500msInterval between sending keep-alives
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME3000countNumber of keep-alive messages per batch

📐 NAT Keep-Alive Operation

VOS3000 NAT Keep-Alive Mechanism:
==================================

Purpose:
========
When devices are behind NAT, the NAT device maintains a mapping table.
If no traffic passes through for a period (typically 30-300 seconds),
the NAT mapping expires, and incoming calls cannot reach the device.

How It Works:
=============
1. Device registers with VOS3000
2. VOS3000 records device IP and port
3. VOS3000 sends periodic keep-alive messages
4. Keep-alive traffic maintains NAT mapping
5. Incoming calls can reach the device

Configuration Example:
======================
SS_SIP_NAT_KEEP_ALIVE_MESSAGE = "HELLO"
SS_SIP_NAT_KEEP_ALIVE_PERIOD = 30 (seconds)

VOS3000 sends "HELLO" to registered devices every 30 seconds.

Important Notes:
================
- If SS_SIP_NAT_KEEP_ALIVE_MESSAGE is empty, keep-alive is disabled
- Period should be less than NAT device timeout (typically 60 seconds)
- For large deployments, adjust SEND_INTERVAL and SEND_ONE_TIME

Usage Scenarios:
================
1. Normal Registration: Device maintains registration via REGISTER
2. Non-REGISTER Devices: VOS3000 sends UDP keep-alive
3. Symmetric NAT: May require media proxy instead

🔧 Session Timer Configuration Guide

ScenarioSS_SIP_SESSION_TTLSS_SIP_NO_TIMER_REINVITE_INTERVALNAT_KEEP_ALIVE_PERIOD
Standard VoIP600 (10 min)7200 (2 hours)30 seconds
Call Center900 (15 min)14400 (4 hours)20 seconds
Wholesale600 (10 min)0 (disabled)30 seconds
Mobile/Unstable300 (5 min)3600 (1 hour)15 seconds

🔧 Configuration Steps

Step-by-Step Session Timer Configuration:
==========================================

1. Navigate to System Parameters:
   Navigation > Operation management > Softswitch management
   > Additional settings > System parameter

2. Configure Session Timer:
   Find: SS_SIP_SESSION_TTL
   Set: 600 (or desired value in seconds)

3. Configure Update Segment:
   Find: SS_SIP_SESSION_UPDATE_SEGMENT
   Set: 2 (refresh interval = TTL/segment)

4. Configure NAT Keep-Alive:
   Find: SS_SIP_NAT_KEEP_ALIVE_MESSAGE
   Set: HELLO (or custom message)

   Find: SS_SIP_NAT_KEEP_ALIVE_PERIOD
   Set: 30 (seconds between keep-alives)

5. Apply Changes:
   Click Apply to save configuration

6. Verify Settings:
   Check CDR for session timeout behavior
   Monitor for 30-second call drops

Important: Changes require softswitch service restart
to take effect in some cases.

🚨 Common Session Timer Problems

📊 Problem Diagnosis Table

SymptomPossible CauseSolution
Calls drop at 30 secondsNAT binding timeout, SIP ALG issueDisable SIP ALG, increase NAT keep-alive
Calls drop at specific intervalsSession timer negotiation failureCheck session timer support, adjust TTL
No incoming calls after idleNAT binding expiredEnable NAT keep-alive, reduce period
Session timer errors in traceEndpoint doesn’t support RFC 4028Use SS_SIP_NO_TIMER_REINVITE_INTERVAL
Re-INVITE rejected by endpointEndpoint doesn’t support re-INVITETry UPDATE method, check endpoint config

🔧 Troubleshooting Session Timer Issues (VOS3000 SIP Session Timer)

Session Timer Troubleshooting Checklist:
=========================================

1. Check Debug Trace:
   System > Debug trace > Enable
   Look for re-INVITE or UPDATE messages
   Check for 200 OK responses

2. Verify Endpoint Support:
   - Check if endpoint includes "timer" in Supported header
   - Look for Session-Expires in INVITE/200 OK
   - Verify endpoint responds to session refresh

3. Check NAT Configuration:
   - Verify NAT keep-alive is enabled
   - Check SS_SIP_NAT_KEEP_ALIVE_PERIOD
   - Monitor for NAT binding expiration

4. Analyze CDR:
   - Check termination reason for session timeouts
   - Look for patterns in call drop timing
   - Compare with session timer configuration

5. Test Different Scenarios:
   - Test calls from different networks
   - Test with different endpoints
   - Test with/without media proxy

Common Fixes:
=============
- Increase SS_SIP_SESSION_TTL for longer refresh intervals
- Reduce SS_SIP_NAT_KEEP_ALIVE_PERIOD for aggressive keep-alive
- Disable SIP ALG on routers
- Enable media proxy for NAT scenarios

📊 Session Timer vs NAT Keep-Alive (VOS3000 SIP Session Timer)

Understanding the difference between session timer and NAT keep-alive is important for proper configuration:

AspectSession TimerNAT Keep-Alive
PurposeDetect failed calls, prevent zombie callsMaintain NAT bindings for incoming calls
ProtocolSIP re-INVITE/UPDATEUDP packets or SIP messages
DirectionBoth directions (refresh negotiation)Server to client (keep binding active)
Default Interval600 seconds (10 minutes)30 seconds
When ActiveDuring active callDuring registration period
RFC ReferenceRFC 4028NAT traversal best practices

❓ Frequently Asked Questions

What happens if both endpoints don’t support session timer?

VOS3000 uses SS_SIP_NO_TIMER_REINVITE_INTERVAL to limit maximum call duration. This prevents zombie calls even when endpoints don’t support RFC 4028. Set this value based on your business needs (default is 7200 seconds / 2 hours).

Why are my calls dropping at exactly 30 seconds?

30-second call drops are typically caused by NAT binding timeouts, not session timer issues. Check if SIP ALG is enabled on your router (should be disabled), and verify NAT keep-alive is configured correctly with a period less than 30 seconds.

Should I use re-INVITE or UPDATE for session refresh?

VOS3000 automatically negotiates the refresh method based on endpoint capabilities. UPDATE is generally preferred as it doesn’t affect SDP negotiation. Both methods work for session timer purposes – VOS3000 handles this automatically.

What is a good SS_SIP_SESSION_TTL value?

The default of 600 seconds (10 minutes) works well for most scenarios. For mobile or unstable networks, consider reducing to 300 seconds (5 minutes) for faster detection of failed calls. For stable enterprise environments, 900 seconds (15 minutes) reduces overhead.

How do I know if NAT keep-alive is working?

Enable debug trace and look for periodic messages matching your SS_SIP_NAT_KEEP_ALIVE_MESSAGE content (default “HELLO”). You should see these messages at intervals matching SS_SIP_NAT_KEEP_ALIVE_PERIOD.

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🌐 Website: www.vos3000.com
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VOS3000 2.1.9.07 Notas de la Version

VOS3000 2.1.9.07 Notas de la Versión – Actualización Completa desde 2.1.8.05 – Important

VOS3000 2.1.9.07 Notas de la Versión – Actualización Completa desde 2.1.8.05

Este documento contiene las Notas oficiales y verificadas de VOS3000 2.1.9.07, preparadas tras una comparación técnica detallada entre los manuales de las versiones 2.1.8.05 y 2.1.9.07. Cada nuevo módulo, lógica de enrutamiento, mejora de facturación, optimización SIP, función de seguridad y actualización de arquitectura backend ha sido documentada completamente.

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)

Para tutoriales técnicos avanzados visite: VOS3000 Technical Blog

Table of Contents


1. Nuevas Secciones Principales y Módulos Funcionales

1.1 Modify CDR (Gestión de Cuenta – 2.4.7)

  • Corrección posterior al cobro
  • Modificación manual de CDR histórico
  • Control administrativo de ajustes de facturación
  • Acceso basado en permisos

Propósito: Permite corrección de facturación sin manipulación directa de base de datos. VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


1.2 Geofencing (Gestión Operativa – 2.5.7)

Reemplaza el módulo antiguo “Prohibited Media IP”.

Nuevas Capacidades:

  • Definición de rango IP (IP inicial + cantidad)
  • Verificación de IP de señalización
  • Verificación de IP en SDP
  • Verificación de IP real RTP

Modos Disponibles:

  • Ignorar
  • Prohibido
  • Permitir

Aplicable en:

  • Nivel global
  • Gateway de enrutamiento
  • Gateway de mapeo
  • Gestión de teléfonos

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


1.3 Escenarios Funcionales (Nuevo Capítulo 3)

3.1 Primer Uso

Escenario actualizado de implementación rápida para despliegue mayorista.

3.2 Pickup Call Transfer

  • Configuración de código de acceso
  • Definición de posición
  • Configuración de asociación
  • Explicación lógica funcional

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


1.4 Explicación de Funciones (Nuevo Capítulo 4.1)

Network Routing Quality Reserve Time

  • SS_GATEWAY_QUALITY_RESERVE_SEPARATE
  • SS_GATEWAY_QUALITY_RESERVE_TIME

Permite cálculo segmentado de ASR/ACD.

NAT Keep

Lógica UDP keep-alive para mantener enlaces NAT.

SIP Timer Protocol

Soporte de session timer y parámetros asociados.

Signaling QoS

  • SS_QOS_SIGNAL
  • SS_QOS_RTP

Control DSCP para paquetes SIP y RTP.

Enable Bilateral Reconciliation

Conciliación en tiempo real entre dos plataformas VOS con alerta de desviación. VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


2. Mejoras de Seguridad y Antifraude

2.1 Motor Dinámico de Lista Negra

  • Detección de límite de llamadas concurrentes
  • Detección de frecuencia maliciosa
  • Detección de ataques sin respuesta
  • Análisis por ventana de tiempo
  • Expiración automática de blacklist
  • Lógica dinámica de bloqueo
  • Parámetros de límite concurrente
  • Intervalo de verificación
  • Temporizador de expiración

2.2 Controles de Autenticación

  • Límite máximo de intentos
  • Suspensión automática tras fallos
  • Protección contra fuerza bruta

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


3. Integración en Tiempo Real y Control Externo

3.1 Reporte HTTP del Estado de Llamada

  • Reporte HTTP en tiempo real
  • IP configurable
  • Puerto configurable
  • Mecanismo de reintento
  • Control de intervalo

3.2 Servidor SIP Redirect Externo (3xx)

  • Servidor externo de decisión de ruta
  • Integración con redirección SIP 3xx
  • Disponibilidad selectiva por teléfono

3.3 Phone Service Layer

  • Reporte online/offline
  • IP y puerto dedicados
  • Redirección automática si offline
  • Monitoreo de estado

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


4. Mejoras en Manejo de Llamadas y Transferencias

4.1 Controles Avanzados de Transferencia

  • Transferencia ciega
  • Transferencia atendida
  • Timeout de acceso en espera
  • Passthrough de ring remoto
  • Cancelación de transferencia
  • Finalización de transferencia
  • Personalización de visualización

4.2 Tono Auxiliar

  • Reproducción local de ringback
  • SS_AUXILIARY_RING_TONE_ACTIVATION_DELAY

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


5. Mejoras en Enrutamiento y Gateway

5.1 Cálculo de Calidad en Tiempo Real

  • SS_GATEWAY_QUALITY_CALCULATE

5.2 Estrategia de Enrutamiento

  • Ordenar por ASR
  • Ordenar por menor tarifa por segundo

5.3 Controles de Conciliación

  • SERVER_GATEWAY_ROUTE_BILATERAL_RECONCILIATION_LINE
  • SERVER_GATEWAY_BILATERAL_RECONCILIATION_PERIOD

5.4 Caller Number Pool

  • Habilitar pool
  • Concurrencia por número
  • Pool de reenvío
  • Control multiplex

5.5 Limitación de Señalización

Control de tasa por gateway.

5.6 Mejoras SIP

  • Control de códigos de respuesta
  • Modo de dirección de respuesta
  • Selección de dirección de solicitud
  • Control G729 annexb
  • Control G723 annexa
  • Soporte 100rel
  • Retry-After
  • Reason Header
  • user=phone
  • Allow Publish
  • Dominio local
  • Forward signal
  • SIP OPTIONS check
  • Ptime adaptativo
  • NAT media SDP first
  • Headers personalizados
  • Privacy support
  • Mapeo de fallos
  • Modo remote ringback

5.7 Manejo LRN

  • Longitud de prefijo
  • Acción ante fallo
  • Enrutamiento por número
  • Prefijo interestatal
  • Prefijo indeterminado

5.8 Mejoras H.323

  • Q.931 ProgressIndicator
  • Selección campo caller
  • Selección campo callee
  • CallProceeding inmediato
  • Conversión Trying
  • Conversión 183(SDP)

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


6. Mejoras CDR y Reportes

6.1 Nuevos Campos CDR

  • Incoming caller
  • Outgoing caller
  • PDD
  • Continue duration
  • Método de facturación
  • Duración paquete
  • Cargos paquete
  • Motivo transparente de colgado

6.2 Reorganización de Análisis

  • Análisis Mapping Gateway
  • Análisis Routing Gateway
  • Análisis rendimiento
  • Análisis llamadas
  • Análisis fallos
  • Análisis diario
  • Análisis por área
  • Análisis cruzado
  • Análisis global

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


7. Mejoras de Facturación

7.1 Customer Package – Suite Order System

  • Paquetes por suscripción
  • Control vigencia
  • Prioridad
  • Minutos gratis
  • Monto gratis
  • Consumo mínimo
  • Renta porcentual
  • Reglas renovación
  • Modo fallo

7.2 Control de Precisión

  • Precisión tarifa
  • Precisión unidad
  • Precisión hold-time
  • Prevención overdraft
  • Fórmula beneficio
  • Prefijo gateway billing
  • Lógica forward billing

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


8. Alarmas y Monitoreo

  • Notificación por voz
  • Estadística pérdida RTP

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


9. Actualización Mayor Backend – Arquitectura Linux 64 Bits

Hasta 2.1.8.05 el sistema era 32 bits.

Limitaciones 32 bits:

  • Límite ~4GB RAM
  • Escalabilidad limitada
  • Menor estabilidad alta concurrencia

Mejoras 2.1.9.07:

  • Arquitectura completa 64 bits
  • Soporte 32GB / 64GB / 128GB+
  • Mejor uso multi-core
  • Caching optimizado
  • Mayor capacidad CPS
  • Mejor eficiencia memoria
  • Mayor estabilidad tráfico mayorista

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


📊 Tabla Comparativa Completa – VOS3000 2.1.8.05 vs 2.1.9.07

Módulo / FunciónVOS3000 2.1.8.05VOS3000 2.1.9.07
Arquitectura BackendLinux 32 bitsLinux 64 bits (Soporte RAM Alta)
Modify CDR (Corrección Post-Facturación)No DisponibleDisponible
Geofencing (Control IP Avanzado)Prohibited Media IP BásicoGeofencing Completo (Señalización + SDP + RTP)
Motor Dinámico de Lista NegraNoSí (Detección Automática)
Detección de Concurrencia de CallerNo
Detección de Ataque sin RespuestaNo
Protección de Reintento de AutenticaciónBásicaAvanzada con Suspensión Automática
Reporte HTTP de Estado de LlamadaNoSí (Push API en Tiempo Real)
Servidor SIP Redirect Externo (3xx)No
Phone Service LayerNoSí (Monitoreo Online/Offline)
Cálculo de Calidad de EnrutamientoEstáticoASR/ACD en Tiempo Real
Conciliación BilateralNo
Caller Number PoolNo
Limitación de SeñalizaciónNo
Protocolo SIP TimerLimitadoMejorado
Soporte SIP 100relNo
Header Retry-AfterNo
Inyección de Header ReasonNo
Soporte Privacy HeaderBásicoMejorado
Manejo Avanzado LRNLimitadoPrefijos y Enrutamiento Mejorado
H.323 ProgressIndicatorNo
Controles Avanzados de TransferenciaBásicosCiega + Atendida + Cancelación + Personalización
Tono AuxiliarNo
Campos CDR ExtendidosLimitadosCampos Ampliados (PDD, Paquetes, etc.)
Análisis Estructurado CDRBásicoGateway + Área + Rendimiento
Customer Package (Suite Order System)No
Control de Precisión de FacturaciónLimitadoParámetros Avanzados
Lógica de Fórmula de BeneficioBásicaMejorada
Soporte de Alarma por VozNo
Estadísticas de Pérdida RTPNo
Soporte de RAM AltaLimitado (~4GB)32GB / 64GB / 128GB+
Estabilidad Alta CPSModeradaAlto Rendimiento

❓ Preguntas Frecuentes (FAQ) – VOS3000 2.1.9.07

1. ¿Cuál es la mejora más importante en VOS3000 2.1.9.07?

La actualización más significativa es la migración completa a arquitectura Linux de 64 bits, permitiendo mayor utilización de memoria RAM, mejor manejo de alta concurrencia y mayor estabilidad para entornos mayoristas de alto tráfico.

2. ¿VOS3000 2.1.9.07 permite optimización de rutas en tiempo real?

Sí. Incorpora cálculo dinámico basado en ASR y ACD que permite ordenar gateways automáticamente según rendimiento en tiempo real.

3. ¿Para qué sirve la función Modify CDR?

Permite modificar registros históricos de facturación sin intervención directa en la base de datos, mejorando la seguridad operativa y la flexibilidad de corrección financiera.

4. ¿Cómo mejora la seguridad el nuevo sistema Geofencing?

Verifica IP de señalización, IP SDP y IP RTP real, pudiendo permitir, bloquear o ignorar rangos IP definidos. Reduce significativamente fraudes VoIP.

5. ¿Incluye esta versión protección antifraude?

Sí. Integra un motor dinámico de blacklist con detección de concurrencia, frecuencia maliciosa, ataques sin respuesta y expiración automática.

6. ¿Puede integrarse con CRM o sistemas externos de facturación?

Sí. Mediante reporte HTTP de estado de llamada y servidor SIP Redirect externo, permite integración en tiempo real con plataformas externas.

7. ¿Soporta conciliación bilateral?

Sí. Dos plataformas VOS pueden realizar conciliación en tiempo real con alertas de desviación.

8. ¿Mejora la interoperabilidad SIP?

Sí. Añade soporte 100rel, Retry-After, Reason header, mejoras NAT, SIP Timer Protocol y control avanzado de encabezados.

9. ¿Qué mejoras de facturación incluye?

Incluye Suite Order System con paquetes de suscripción, minutos gratis, consumo mínimo, renta porcentual y control avanzado de precisión.

10. ¿Es adecuada para tráfico mayorista de alto volumen?

Sí. Gracias a la arquitectura 64 bits, motor antifraude, optimización de rutas y mejor uso de memoria, ofrece mayor estabilidad frente a versiones 2.1.8.x.

VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


Recursos Oficiales

1. VOS3000 Technical Blog: https://www.vos3000.com/blog/vos3000
2. Official Blog: https://www.vos3000.com/blog/
3. Downloads & Manuals: https://www.vos3000.com/downloads.php
4. Advanced Security Guides: https://multahost.com/blog/


Contacto para Hosting VOS3000

Instalación única / Hosting dedicado disponible.

WhatsApp: +8801911119966

Enlace directo: https://wa.me/+8801911119966

Servidores disponibles en China | Hong Kong | Vietnam | Tailandia – VOS3000 2.1.9.07 Notas de la Version (generated by AI by using both version user manual)


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