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VOS3000 parameter description, VOS3000 system parameter, VOS3000 data maintenance, VOS3000 data report, VOS3000 number management

VOS3000 Parameter Description: Complete Configuration Reference Guide Free

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VOS3000 Parameter Description: Complete Configuration Reference Guide

VOS3000 parameter description is the most comprehensive technical reference available for VoIP system administrators who need to configure and optimize their softswitch installations. This complete configuration reference guide covers every single parameter available in VOS3000 version 2.1.9.07, organized into logical categories for easy navigation and practical implementation. Whether you are managing a small wholesale VoIP operation or a large-scale telecom infrastructure, understanding these parameters is essential for achieving optimal call quality, billing accuracy, and system reliability. Based on the official VOS3000 2.1.9.07 manual (Section 4.3.5, Pages 222-252), this guide provides detailed explanations of each parameter including default values, valid ranges, and practical usage scenarios.

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Table of Contents

🔍 What is VOS3000 Parameter Description

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5 (Pages 222-252)

The VOS3000 parameter description framework organizes all configuration settings into a hierarchical structure that reflects the functional architecture of the softswitch system. At the highest level, parameters are divided into three primary categories: VOS3000 server parameters, softswitch parameters (including H323, SIP, and system subcategories), and audio service parameters. Each category controls specific aspects of system behavior, and understanding these categories is crucial for effective system administration. The VOS3000 softswitch platform contains over 200 configurable parameters that control every aspect of system behavior, from billing precision and alarm thresholds to SIP timer values and media proxy settings.

📊 VOS3000 Parameter Description Categories

📁 Category📋 Description📖 Manual Pages
VOS3000 ParametersServer-level parameters for billing, alarms, reports, security222-228
Softswitch H323 ParametersH.323 protocol settings for gateway communications229-230
Softswitch SIP ParametersSIP protocol settings including NAT, timers, authentication230-237
Softswitch System ParametersCore softswitch settings for media, calls, endpoints237-239
Audio Service ParametersIVR, voicemail, callback service settings239-241

⚙️ How to Access VOS3000 Parameter Description Settings

Accessing the VOS3000 parameter description settings requires navigating through the VOS3000 client interface to the appropriate configuration menus. For server parameters, administrators should navigate to System Management, then select System Parameter to view and modify the parameter list. For softswitch parameters including H323, SIP, and system subcategories, the path is Operation Management followed by Softswitch Management, then Additional Settings, and finally System Parameter. Audio service parameters are accessed through the audio service configuration interface.

📍 Navigation Paths for Parameter Access

StepNavigation PathAction
1System ManagementExpand navigation tree
2System ParameterDouble-click to open parameter table
3Operation Management > Softswitch ManagementSelect softswitch node
4Additional SettingsRight-click → Additional settings
5System Parameter TabFind and modify parameters
6Apply ChangesClick OK to save modifications

📋 VOS3000 Server Parameters Complete List

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.1 (Pages 222-228)

The VOS3000 parameter description for server parameters encompasses all configuration settings that control the core server functionality of the softswitch platform. These parameters determine how the server handles billing calculations, generates reports, manages alarms, interacts with databases, and enforces security policies. Server parameters are prefixed with “SERVER_” in the parameter name, making them easily identifiable in the configuration interface.

🔔 Alarm Configuration Parameters in VOS3000

Alarm configuration parameters within the VOS3000 parameter description control how the system monitors and reports various operational conditions. These parameters define thresholds for generating alerts, specify notification methods, and configure alarm suppression settings. Proper configuration of alarm parameters ensures that administrators receive timely notifications of critical system conditions without being overwhelmed by excessive alerts.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SERVER_ALARM_CUSTOMER_BALANCE_MAX_SIZE1000Number of accounts in Balance Alarm settings menu223
SERVER_ALARM_DATABASE_IGNORE_ERROR_CODE–Database error codes to ignore without triggering warnings223
SERVER_ALARM_DISABLEOffOff enables alarm system, On disables all alarms223
SERVER_ALARM_E164SDefaultDefault E164 number for Alarm Management223
SERVER_ALARM_EMAILDefaultDefault email address for alarm notifications223
SERVER_ALARM_EMAIL_DELAY300Interval in seconds between email alarm notifications223
SERVER_ALARM_ENABLE_EMAILOffEnable email alarm notifications (On/Off)223
SERVER_ALARM_ENABLE_VOICEOffEnable voice call alarm notifications (On/Off)223

💰 Billing System Parameters in VOS3000 Parameter Description

The billing system parameters form a critical component of the VOS3000 parameter description because they directly affect revenue calculation and financial accuracy. These parameters control billing precision, fee calculation methods, free call duration settings, and various billing behaviors that determine how calls are charged. Misconfiguration of billing parameters can result in revenue loss, customer disputes, or billing errors.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SERVER_BILLING_FEE_PRECISION0.0000000Billing money accuracy precision (0-1000 decimal places)224
SERVER_BILLING_FEE_UNIT0.0000000Billing money unit for charge calculations (0-1000)224
SERVER_BILLING_FORWARD_PREFIX–Billing prefix for Call Transfer scenarios224
SERVER_BILLING_FREE_E164S–Service numbers for free calls with no time limit224
SERVER_BILLING_FREE_TIME0Free duration in seconds to deduct from charged time224
SERVER_BILLING_GATEWAY_ROUTE_PREFIX–Routing gateway additional prefix for billing224
SERVER_BILLING_HOLD_TIME_PRECISION1000Time precision in milliseconds for billing duration224
SERVER_BILLING_NO_CDR_E164S–Numbers that will not create CDR records224
SERVER_BILLING_PREVENT_OVERDRAFT_ADVANCE_TIME1Account anti-overdraft advance minutes (1-15)224
SERVER_BILLING_PROFIT_CALCULATECall charges – Sub – Call expenseFormula for call profit calculation224

📊 CDR and Reporting Parameters

Call Detail Record (CDR) and reporting parameters within the VOS3000 parameter description govern how call records are generated, stored, and processed for reporting purposes. These parameters determine CDR file formats, storage intervals, queue sizes, and automatic report generation settings. Proper configuration of CDR parameters is essential for maintaining accurate call records and enabling detailed traffic analysis.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SERVER_CDR_FILE_WRITE_INTERVALNoneInterval in seconds for creating new CDR files (60-86400)225
SERVER_CDR_FILE_WRITE_MAX2048Maximum number of CDR files to retain (10-4096)225
SERVER_CDR_REAL_TIME_REPORT_SERVER–Address for real-time CDR reporting server225
SERVER_MAX_CDR_PENDING_LIST_LENGTH100000Maximum length of CDR processing queue (10000-100000)225
SERVER_QUERY_CDR_DENY_TIME–Hours when CDR query is denied (e.g., 18,19,20,21)225
SERVER_QUERY_CDR_MAX_DAY_INTERVAL31Maximum days for CDR query interval225

📈 Automatic Report Generation Parameters

The VOS3000 parameter description includes numerous parameters that control automatic report generation for business intelligence and operational analysis purposes. These reports are generated daily at approximately 1:00 AM and include revenue reports, gateway billing analysis, clearing reports, and various analytical reports.

⚙️ Parameter Name📊 Default📝 Report Generated
SERVER_REPORT_AGENT_INCOMEOnAgent Income Report
SERVER_REPORT_CLEARING_CUSTOMER_FEEOffClearing Account Details Report
SERVER_REPORT_CUSTOMER_FEEOnRevenue Details Report
SERVER_REPORT_GATEWAY_FEEOnGateway Bill Report
SERVER_REPORT_PHONE_FEEOnPhone Bill Report
SERVER_REPORT_GATEWAY_ROUTING_LOCATION_ASR_ACDOnRouting Gateway Area Analysis Report

🔒 Security and Authentication Parameters

Security parameters in the VOS3000 parameter description establish the foundational security posture of the softswitch system. These parameters control password policies, login attempt restrictions, session management, and various authentication behaviors that protect the system from unauthorized access. In today’s threat landscape where VoIP systems are frequent targets for fraud and abuse, proper configuration of security parameters is essential.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SERVER_LOGIN_FAILED_DISABLE_TIME120Seconds to disable login after failed attempts (30-7200)226
SERVER_PASSWORD_LENGTH8Default minimum password length requirement226
SERVER_PASSWORD_TERMINAL_ADDITIONAL_CHARACTERS–Additional characters for phone/gateway random passwords226
SERVER_VERIFY_CLEARING_CUSTOMEROffVerify clearing account balance against minimum limit226
SERVER_VERIFY_CLEARING_CUSTOMER_REMAIN_MONEY_LIMIT0.0Clearing account minimum balance limit (0-10000000)226

🖥️ System Configuration Parameters

System configuration parameters in the VOS3000 parameter description control various operational aspects of the server including NTP time synchronization, display settings, database version management, and network configuration. These parameters establish the operational environment in which the softswitch functions.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SERVER_NTP_SERVERtime-a.nist.govNetwork time server (SNTP) for system time sync227
SERVER_DATABASE_VERSION–Current database version identifier227
SERVER_DISPLAY_MONEY_PRECISION3Money display precision (e.g., 3 shows 1.000)227
SERVER_DNS_UPDATE_INTERVAL600DNS update interval in seconds for Domain Management227
SERVER_SOFTSWITCH_CLUSTER–IP list of softswitch cluster nodes227
SERVER_QUERY_MAX_SIZE30000000Maximum data query limit in items227
SERVER_QUERY_ONE_PAGE_SIZE10000Number of data items per query page227
SERVER_TRACE_FILE_LENGTH40960Debug file size in KB227

📡 Softswitch H323 Parameters in VOS3000 Parameter Description

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Pages 229-230)

The H323 parameters within the VOS3000 parameter description control the behavior of H.323 protocol signaling for gateway communications. H.323 is an ITU-T standard protocol suite for multimedia communications over packet-based networks, and it remains widely deployed in enterprise and carrier VoIP environments despite the growing adoption of SIP.

⚙️ Parameter Name📊 Default📝 Description📖 Page
SS_H245_PORT_RANGE10000,39999H245 port range for media control channels229
SS_H323_DTMF_METHODH.245 alphanumericDefault DTMF transmission mode for H.323229
SS_H323_NUMBERING_PLANUnknownPlan(0)Default numbering plan in Routing Gateway H323229
SS_H323_NUMBER_TYPEUnknownType(0)Default number type in Routing Gateway H323229
SS_H323_TIMEOUT_ALERTING120Alerting timeout in seconds for Routing Gateway H323230
SS_H323_TIMEOUT_SETUP5Setup timeout in seconds for H.323 call establishment230

📞 Softswitch SIP Parameters Complete Reference

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Pages 230-237)

The SIP parameters represent one of the most extensive sections within the VOS3000 parameter description, reflecting the complexity and flexibility of the Session Initiation Protocol. SIP has become the dominant signaling protocol for VoIP communications, and VOS3000 provides comprehensive configuration options for controlling every aspect of SIP behavior including authentication, NAT traversal, session timers, and timeout values.

🔑 SIP Authentication Parameters

⚙️ Parameter Name📊 Default📝 Description📖 Page
SS_SIP_AUTHENTICATION_CODE–SIP authentication code for gateway registration230
SS_SIP_AUTHENTICATION_REALM–SIP authentication realm for digest authentication230

📡 NAT Keep-Alive Parameters

NAT keep-alive parameters in the VOS3000 parameter description are critical for maintaining connectivity with endpoints behind NAT devices. These parameters control the message content, sending period, and batching behavior for UDP heartbeat messages that prevent NAT bindings from expiring.

⚙️ Parameter Name📊 Default📏 Range📝 Description
SS_SIP_NAT_KEEP_ALIVE_MESSAGEHELLOText stringContent of NAT keep-alive UDP packet (empty = disabled)
SS_SIP_NAT_KEEP_ALIVE_PERIOD3010-86400 secInterval between keep-alive transmissions
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL5001-10000 msDelay between individual keep-alive packets in batch
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME30001-10000Number of keep-alive packets sent per batch cycle

⏱️ SIP Session Timer Parameters

Session timer parameters in the VOS3000 parameter description control the SIP session timer functionality that prevents “zombie calls” from persisting in the system. Based on RFC 4028, the session timer mechanism ensures that failed or hung calls are detected and cleaned up automatically.

⚙️ Parameter Name📊 Default📏 Range📝 Description
SS_SIP_SESSION_TTL60060-86400 secDetecting SIP connected status interval (Session-Expires)
SS_SIP_SESSION_UPDATE_SEGMENT22-10Divisor for refresh interval calculation (TTL/segment)
SS_SIP_SESSION_MIN_SE9090-3600 secMinimum session expires value per RFC 4028
SS_SIP_NO_TIMER_REINVITE_INTERVAL72000-86400 secMaximum call duration for non-timer endpoints

🎛️ Softswitch System Parameters in VOS3000 Parameter Description

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Pages 237-239)

Softswitch system parameters control core softswitch functionality including media handling, call processing, gateway management, and blacklist/whitelist behavior. These parameters affect how the softswitch processes calls and interacts with gateways and endpoints.

🎬 Media and Call Processing Parameters

⚙️ Parameter Name📊 Default📝 Description📖 Page
SS_MEDIA_PROXY_MODE0Media proxy mode (0=disabled, 1=enabled)237
SS_MEDIA_PROXY_PORT_RANGE40000,59999Port range for media proxy RTP traffic237
SS_MAX_CALL_DURATION0Maximum call duration in seconds (0=unlimited)237
SS_ENDPOINT_EXPIRE3600Terminal registration expiry time in seconds237
SS_GATEWAY_ASR_RESERVE_TIME600ASR reserve time for gateway in seconds238
SS_GATEWAY_ACD_RESERVE_TIME600ACD reserve time for gateway in seconds238

🚫 Dynamic Black List Parameters

⚙️ Parameter Name📊 Default📝 Description
SS_BLACK_LIST_CALLER_MALICIOUS_CALL_LIMIT1000Max calls triggering malicious call blocking
SS_BLACK_LIST_CALLER_MALICIOUS_CALL_EXPIRE3600Duration for malicious call block in seconds
SS_BLACK_LIST_NO_ANSWER_LIMIT100Consecutive no-answer calls triggering block
SS_BLACK_LIST_NO_ANSWER_EXPIRE3600Duration for no-answer block in seconds

🎵 Audio Service Parameters in VOS3000 Parameter Description

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.3 (Pages 239-241)

Audio service parameters control the IVR (Interactive Voice Response) system, voicemail functionality, callback services, and other value-added audio features in VOS3000. These parameters determine codec priorities, language settings, timeout values, and session behavior for audio services.

⚙️ Parameter Name📊 Default📝 Description📖 Page
IVR_CODEC_PRIORITYG.711A,G.711U,G.729,G.723Codec priority for IVR media239
IVR_DEFAULT_LANGUAGEenDefault language for IVR prompts239
IVR_MEDIA_CHECK_TIME_OUT3000Media check timeout in milliseconds240
IVR_RINGING_TIMEOUT60Ringing timeout in seconds240
IVR_SIP_SESSION_TTL600SIP session TTL for IVR calls240
IVR_VOICEMAIL_MAX_DURATION120Maximum voicemail duration in seconds241

⚙️ VOS3000 Parameter Description Best Practices

Implementing effective VOS3000 parameter description management requires adherence to established best practices that minimize risk and ensure system stability. The following recommendations are derived from extensive deployment experience and reflect industry-standard approaches to configuration management.

📋 Change Management Recommendations

  • Document current settings: Before making any changes, record the current parameter value and description for rollback reference.
  • Research parameter function: Review the parameter description in the interface and consult the VOS3000 manual to fully understand the parameter’s purpose.
  • Test before production: Always test parameter changes in a non-production environment before applying to production systems.
  • Apply changes during maintenance windows: Plan parameter changes during periods when temporary service interruption is acceptable.
  • Verify after changes: Confirm that parameter changes produce the expected behavior and do not cause unintended side effects.

🔧 Parameter Optimization Tips

🏢 Scenario⏱️ SESSION_TTL📡 NAT_PERIOD🚫 MAX_DURATION
Standard VoIP Wholesale600 (10 min)30 sec0 (unlimited)
Call Center Operations900 (15 min)20 sec14400 (4 hrs)
Mobile/Unstable Networks300 (5 min)15 sec3600 (1 hr)
Enterprise PBX1200 (20 min)30 sec28800 (8 hrs)

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📦 Service📝 Description💼 Includes
VOS3000 InstallationComplete server setupOS, VOS3000, Database, Security
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❓ Frequently Asked Questions about VOS3000 Parameter Description

What is the most important VOS3000 parameter description for billing accuracy?

The SERVER_BILLING_FEE_PRECISION and SERVER_BILLING_FEE_UNIT parameters are critical for billing accuracy. These parameters control the decimal precision and billing unit for charge calculations. Configure these parameters according to your business requirements and regulatory requirements for billing precision.

How do I enable NAT keep-alive in VOS3000 parameter description?

To enable NAT keep-alive, set SS_SIP_NAT_KEEP_ALIVE_MESSAGE to a non-empty value (default is “HELLO”). If this parameter is empty, NAT keep-alive is disabled. Configure SS_SIP_NAT_KEEP_ALIVE_PERIOD to control the interval between keep-alive transmissions (default is 30 seconds).

What happens if I set SS_SIP_SESSION_TTL too low?

Setting SS_SIP_SESSION_TTL too low (below 90 seconds) may cause frequent session refresh messages, increasing network traffic and potentially causing call quality issues. The minimum recommended value is 90 seconds as specified in RFC 4028. Values below this may trigger “422 Session Interval Too Small” errors from endpoints.

How do I disable automatic report generation?

To disable automatic generation of specific reports, set the corresponding SERVER_REPORT_ parameter to “Off” in the System Parameter interface. For example, to disable the Agent Income Report, set SERVER_REPORT_AGENT_INCOME to “Off”. Disabled reports can still be generated manually through the client interface.

Can I use VOS3000 parameter description to limit maximum call duration?

Yes, use the SS_MAX_CALL_DURATION parameter to limit the maximum call duration for all calls. Set the value in seconds (0 means unlimited). This parameter is useful for preventing runaway calls and controlling costs. Individual accounts may have additional duration limits configured in their settings.

Where can I get help with VOS3000 parameter description configuration?

MultaHost provides comprehensive technical support for VOS3000 parameter description configuration. Our experienced team can assist with parameter selection, configuration best practices, and troubleshooting. For immediate assistance, contact us via WhatsApp at +8801911119966. Additional resources are available at vos3000.com/downloads.php.

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

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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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📦 Service📝 Description💼 Includes
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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 QoS Configuration: Complete Voice Quality Optimization Guide

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VOS3000 QoS Configuration: Complete Voice Quality Optimization Guide

VOS3000 QoS configuration is essential for ensuring superior voice quality in enterprise and carrier VoIP deployments. By properly marking SIP signaling and RTP media packets with DSCP (Differentiated Services Code Point) values, VOS3000 enables network infrastructure to prioritize voice traffic, reducing latency, jitter, and packet loss that degrade call quality. This comprehensive guide covers all QoS features based on official VOS3000 2.1.9.07 documentation.

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🔍 Understanding VoIP QoS

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.4 (Page 214)

Quality of Service (QoS) in VoIP refers to the ability to prioritize voice traffic over data traffic on IP networks. Voice calls are highly sensitive to network conditions – even small amounts of latency, jitter, or packet loss can significantly degrade call quality. QoS mechanisms ensure voice packets receive preferential treatment.

📊 Voice Quality Requirements

MetricVoice RequirementImpact if ExceededQoS Benefit
Latency< 150ms one-wayEcho, talk-over, delayed responsePriority queuing reduces delay
Jitter< 30ms variationChoppy audio, robotic voiceConsistent queuing reduces variation
Packet Loss< 1%Clicks, pops, missing syllablesPriority treatment reduces drops
Bandwidth~30-90 kbps per callCongestion, quality degradationGuaranteed bandwidth allocation

⚙️ VOS3000 QoS Parameters

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.4 (Page 214)

📊 VOS3000 QoS Configuration Parameters

ParameterDefaultDescriptionApplies To
SS_QOS_SIGNAL0xa0QoS marking for SIP signaling packetsSIP INVITE, REGISTER, BYE, etc.
SS_QOS_RTP0xa0QoS marking for RTP media packetsVoice/audio RTP streams

📐 Understanding DSCP Values

The QoS parameters use hexadecimal values that correspond to the DSCP field in the IP header:

Hex ValueBinaryDSCP NameTypical UsePriority Level
0xb8101110EF (Expedited Forwarding)Voice RTPHighest
0xa0101000CS5 (Class Selector 5)Voice SignalingHigh
0x88100010AF41Video ConferencingMedium-High
0x00000000BE (Best Effort)Regular DataDefault

📐 How VOS3000 QoS Works

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.4 (Page 214)

📊 IP Header DSCP Field

IP Header QoS Field Structure:
==============================

The Differentiated Services Field in IP header:

Bits:     0   1   2   3   4   5   6   7
        +---+---+---+---+---+---+---+---+
        |   DSCP (6 bits)   |   ECN     |
        +---+---+---+---+---+---+---+---+

DSCP = Differentiated Services Code Point
ECN  = Explicit Congestion Notification

VOS3000 Default: 0xa0
======================
Binary:     10100000
DSCP bits:  101000 (DSCP 40 = CS5)
ECN bits:   00

This means:
- DSCP Class Selector 5
- High priority for signaling
- No ECN marking

Wireshark Display:
==================
Differentiated Services Field: 0xa0 (DSCP: CS5, ECN: Not-ECT)

📊 VOS3000 QoS Application

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.4 (Page 214)

Packet TypeParameterDefault ValueEffect
SIP SignalingSS_QOS_SIGNAL0xa0 (CS5)Fast call setup, priority for INVITE/REGISTER
RTP MediaSS_QOS_RTP0xa0 (CS5)Clear voice, reduced jitter and loss

🔧 Configuring QoS in VOS3000

📍 Configuration Location

Navigate to: Operation management > Softswitch management > Additional settings > System parameter

⚙️ Configuration Steps (VOS3000 QoS)

Step-by-Step VOS3000 QoS Configuration:
========================================

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

2. Locate QoS Parameters:
   Find: SS_QOS_SIGNAL
   Find: SS_QOS_RTP

3. Set Signaling QoS:
   Parameter: SS_QOS_SIGNAL
   Default: 0xa0 (CS5)
   Options:
   - 0xa0 = CS5 (recommended for SIP signaling)
   - 0x00 = Best Effort (no priority)
   - 0xb8 = EF (if signaling needs highest priority)

4. Set RTP Media QoS:
   Parameter: SS_QOS_RTP
   Default: 0xa0 (CS5)
   Options:
   - 0xb8 = EF (recommended for voice RTP)
   - 0xa0 = CS5 (acceptable for voice)
   - 0x00 = Best Effort (not recommended)

5. Apply Configuration:
   Click Apply to save changes

6. Verify with Packet Capture:
   Use Wireshark to confirm DSCP markings

Recommended Values:
===================
SS_QOS_SIGNAL = 0xa0  (CS5 - High priority signaling)
SS_QOS_RTP    = 0xb8  (EF - Highest priority voice)

📊 Network Configuration for VOS3000 QoS

QoS markings in VOS3000 are only effective if network infrastructure respects them. Here’s how to configure common network devices:

🔹 Cisco Router QoS Configuration

Cisco Router QoS Configuration Example:
========================================

! Define class maps for voice traffic
class-map match-any VOICE-SIGNAL
 match ip dscp cs5

class-map match-any VOICE-RTP
 match ip dscp ef

! Define policy map
policy-map VOICE-POLICY
 class VOICE-RTP
  priority percent 30
  set dscp ef
 class VOICE-SIGNAL
  bandwidth percent 5
  set dscp cs5

! Apply to interface
interface GigabitEthernet0/0
 service-policy output VOICE-POLICY

! Verify configuration
show policy-map interface GigabitEthernet0/0

🔹 MikroTik RouterOS QoS Configuration

MikroTik RouterOS QoS Configuration:
=====================================

# Create mangle rules to mark packets
/ip firewall mangle
add chain=postrouting protocol=udp dst-port=5060 action=mark-packet new-packet-mark=sip-signal passthrough=yes
add chain=postrouting protocol=udp dst-port=10000-20000 action=mark-packet new-packet-mark=voice-rtp passthrough=yes

# Create queue tree for prioritization
/queue tree
add name="voice-rtp" parent=global packet-mark=voice-rtp priority=1 max-limit=10M
add name="sip-signal" parent=global packet-mark=sip-signal priority=2 max-limit=2M

# Verify with packet sniffing
/tool sniffer quick protocol=udp port=5060,10000-20000

🔹 Linux tc QoS Configuration

Linux Traffic Control QoS Example:
===================================

# Create root qdisc
tc qdisc add dev eth0 root handle 1: htb default 20

# Create classes
tc class add dev eth0 parent 1: classid 1:1 htb rate 100mbit
tc class add dev eth0 parent 1:1 classid 1:10 htb rate 30mbit prio 1  # Voice
tc class add dev eth0 parent 1:1 classid 1:20 htb rate 70mbit prio 2  # Data

# Filter by DSCP
tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match ip dscp 0xb8 0xfc flowid 1:10
tc filter add dev eth0 protocol ip parent 1:0 prio 2 u32 match ip dscp 0xa0 0xfc flowid 1:10

# Verify
tc -s qdisc show dev eth0

📊 End-to-End QoS Chain

For effective QoS, all network elements must be configured:

Network ElementConfiguration RequiredImpact if Not Configured
VOS3000 ServerSet SS_QOS_SIGNAL and SS_QOS_RTPPackets sent without priority markings
Local RouterQoS policy matching DSCP valuesVoice packets treated as data
WAN/MPLSProvider respects DSCP or maps to MPLS EXPCongestion causes voice quality issues
Remote RouterQoS policy on egressLast-mile congestion affects quality
EndpointSend/receive marked packetsMay mark differently, causing mismatch

🔍 Verifying QoS Configuration

📊 Wireshark Analysis

Verifying QoS with Wireshark:
=============================

1. Capture packets on VOS3000 server or network

2. Filter for SIP signaling:
   Display filter: sip

3. Filter for RTP media:
   Display filter: rtp

4. Check DSCP field:
   - Expand IP header in packet details
   - Look for "Differentiated Services Field"
   - Verify value matches configuration

Expected Results:
=================
SIP packets: Differentiated Services Field: 0xa0 (DSCP: CS5)
RTP packets: Differentiated Services Field: 0xb8 (DSCP: EF)

Wireshark Column Setup:
=======================
Add "DSCP Value" column to quickly verify markings:
1. Right-click column header
2. Column Preferences
3. Add new column: "DSCP" with type "DSCP Value"

Common Issues to Check:
=======================
- Value shows 0x00 = QoS not applied
- Value doesn't match configuration = Check parameter setting
- Different values on different interfaces = Router rewriting DSCP

📊 QoS Verification Commands

PlatformCommandPurpose
Ciscoshow policy-map interfaceView QoS statistics
MikroTik/queue tree print statsView queue statistics
Linuxtc -s qdisc showView traffic control stats
tcpdumptcpdump -i eth0 -vv ipView DSCP in packet headers

🚨 QoS Troubleshooting

📊 Common QoS Problems

ProblemSymptomSolution
Packets unmarkedWireshark shows DSCP 0x00Verify SS_QOS parameters are set correctly
Router ignoring DSCPVoice quality poor during congestionConfigure QoS policy on router
DSCP rewritingDifferent DSCP on different network segmentsCheck router config for DSCP rewriting rules
Inconsistent markingSome packets marked, some notCheck if media proxy is interfering
WAN provider strips DSCPQoS works locally but not across WANNegotiate QoS with provider, use MPLS EXP

🔧 QoS Troubleshooting Steps

QoS Troubleshooting Checklist:
==============================

1. Verify VOS3000 Configuration:
   ☐ Check SS_QOS_SIGNAL value
   ☐ Check SS_QOS_RTP value
   ☐ Verify parameters applied after change

2. Verify Packet Marking:
   ☐ Capture packets with Wireshark/tcpdump
   ☐ Check DSCP field in IP header
   ☐ Confirm values match configuration

3. Verify Network QoS:
   ☐ Check router QoS configuration
   ☐ Verify DSCP matching rules
   ☐ Check queue statistics for voice traffic

4. Verify End-to-End:
   ☐ Test from endpoint to VOS3000
   ☐ Test through entire network path
   ☐ Check DSCP preservation at each hop

5. Performance Testing:
   ☐ Run voice quality tests under load
   ☐ Compare MOS scores with/without QoS
   ☐ Monitor latency, jitter, packet loss

Best Practices:
===============
- Document your QoS configuration
- Test during peak traffic periods
- Monitor QoS statistics regularly
- Coordinate with WAN providers
- Consider using separate VLAN for voice

📊 MPLS QoS Considerations

For MPLS networks, DSCP values may need to be mapped to MPLS EXP bits:

DSCP ValueMPLS EXPTraffic Type
EF (0xb8)7Real-time voice
CS5 (0xa0)5Voice signaling
AF41 (0x88)4Interactive video
BE (0x00)0Best effort data

❓ Frequently Asked Questions

What DSCP value should I use for RTP voice packets?

The recommended DSCP value for voice RTP is EF (Expedited Forwarding, 0xb8), which provides the highest priority treatment. However, the VOS3000 default is CS5 (0xa0), which is also acceptable for voice. For best results in controlled networks, use 0xb8 for RTP and 0xa0 for SIP signaling.

Does QoS work over the public internet?

No, QoS markings are generally not respected over the public internet. Most ISPs either ignore DSCP values or strip them entirely. QoS is effective only on networks you control (LAN, WAN with SLA, MPLS) or where you have agreement with the provider to honor markings.

Why do my QoS settings seem to have no effect?

QoS requires end-to-end configuration. Check: 1) VOS3000 parameters are set correctly, 2) Network devices are configured to match and prioritize marked packets, 3) There’s actual congestion for QoS to manage, 4) DSCP values aren’t being rewritten by intermediate devices.

Can different endpoints have different QoS settings?

VOS3000 QoS parameters (SS_QOS_SIGNAL and SS_QOS_RTP) apply globally to all calls processed by the softswitch. For per-endpoint QoS differentiation, you would need to implement QoS policies on network devices based on IP addresses or other criteria.

Should signaling and media use the same DSCP value?

Generally, media (RTP) should have higher priority than signaling because it’s more sensitive to delay and jitter. A common approach is EF (0xb8) for RTP and CS5 (0xa0) for SIP signaling. However, VOS3000 defaults both to CS5, which works well in most scenarios.

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

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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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