VOS3000 session timer, VOS3000 call end reasons, VOS3000 Work Calendar, VOS3000 geofencing, VOS3000软交换参数优化, VOS3000错误代码大全, VOS3000账户权限管理

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

April 7, 2026April 7, 2026 king

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.

📞 Need help configuring VOS3000 session timer? WhatsApp: +8801911119966

🔍 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 Failure	Call remains “active” indefinitely in system	Session expires, call terminated cleanly
Network Disconnection	No notification, resources wasted	Refresh fails, session cleaned up
Device Crash	Zombie call persists for hours/days	Maximum session duration enforced
NAT Timeout	One-way audio, confused state	Session refresh detects failure
Billing Impact	Incorrect CDR duration, revenue loss	Accurate 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_TTL	600	60-86400 sec	Detecting SIP connected status interval (Session-Expires value)	230
SS_SIP_SESSION_UPDATE_SEGMENT	2	2-10	Divisor for refresh interval calculation (TTL/segment)	230
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP	0	0-3600 sec	Terminate session before actual timeout (margin)	230
SS_SIP_NO_TIMER_REINVITE_INTERVAL	7200	0-86400 sec	Maximum call duration for non-timer endpoints	230
SS_SIP_SESSION_MIN_SE	90	90-3600 sec	Minimum session expires value per RFC 4028	231

📊 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_MESSAGE	HELLO	Text string	Content of NAT keep-alive UDP packet	212
SS_SIP_NAT_KEEP_ALIVE_PERIOD	30	10-86400 sec	Interval between keep-alive transmissions	212
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL	500	1-10000 ms	Delay between individual keep-alive packets in batch	212
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME	3000	1-10000	Number of keep-alive packets sent per batch cycle	212

🔄 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

Step	Navigation Path	Action
1	Operation management	Click main menu
2	Softswitch management	Select softswitch node
3	Additional settings	Right-click → Additional settings
4	System parameter tab	Find session timer parameters
5	Modify values	Edit desired parameters
6	Apply changes	Click OK to save

⚙️ Recommended Configuration by Deployment Type

🏢 Scenario	⏱️ SESSION_TTL	🔄 SEGMENT	🚫 NO_TIMER_INTERVAL	📡 NAT_PERIOD
Standard VoIP Wholesale	600 (10 min)	2	0 (disabled)	30 sec
Call Center Operations	900 (15 min)	3	14400 (4 hrs)	20 sec
Mobile/Unstable Networks	300 (5 min)	2	3600 (1 hr)	15 sec
Enterprise PBX	1200 (20 min)	2	28800 (8 hrs)	30 sec
High-Security Environment	180 (3 min)	2	1800 (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 seconds	NAT binding timeout, not session timer	Enable NAT keep-alive, reduce period to 15-20s	Page 212
Calls drop at 5-minute intervals	Session refresh failing	Check if endpoint supports re-INVITE/UPDATE	Page 213
“422 Session Interval Too Small” error	Session-Expires below minimum	Increase SS_SIP_SESSION_MIN_SE or TTL	Page 231
No incoming calls after idle period	NAT binding expired	Verify NAT keep-alive is enabled and working	Page 212
Re-INVITE rejected with 491	Glare condition (simultaneous re-INVITEs)	Normal – VOS3000 will retry automatically	Page 213
Zombie calls still occurring	Session timer not negotiated	Check NO_TIMER_REINVITE_INTERVAL setting	Page 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 Purpose	Detect failed calls, prevent zombie sessions	Maintain NAT bindings for incoming calls
RFC Standard	RFC 4028 (SIP Session Timer)	NAT traversal best practices
Protocol Used	SIP re-INVITE or UPDATE messages	UDP packets or SIP messages
When Active	During active call (after 200 OK)	While endpoint is registered
Direction	Bidirectional (negotiated refresh)	Server to endpoint (unidirectional)
Default Interval	600 seconds (10 minutes)	30 seconds
Failure Result	Call terminated, CDR updated	Incoming calls may fail
Endpoint Support Required	Yes (RFC 4028 compliance)	No (transparent to endpoint)

💰 VOS3000 Installation and Support Services

Need professional help with VOS3000 session timer configuration? Our team provides comprehensive VOS3000 services including installation, configuration, and ongoing technical support.

📦 Service	📝 Description	💼 Includes
VOS3000 Installation	Complete server setup	OS, VOS3000, Database, Security
Session Timer Configuration	Optimize for your environment	NAT handling, Timer tuning
Technical Support	24/7 remote assistance	Troubleshooting, Debug, Analysis

📞 Contact us for VOS3000: WhatsApp: +8801911119966

❓ 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.

📞 Get Expert VOS3000 Session Timer Support

Need assistance configuring VOS3000 session timer or troubleshooting call drop issues? Our VOS3000 experts provide comprehensive support for session management, NAT traversal, and VoIP infrastructure optimization.

📱 WhatsApp: +8801911119966

📞 Need Professional VOS3000 Setup Support?

For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:

📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads

Negocio VoIP Mayorista, VICIDIAL Servidor, Softswitch Barato, VoIP批发业务, 软交换比较, VOS3000 session timer, VOS3000 call end reasons, VOS3000 Work Calendar, VOS3000 geofencing, VOS3000软交换参数优化, VOS3000错误代码大全, VOS3000账户权限管理

SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons

VOS3000 Media Proxy and System Parameters: Complete Important Configuration Reference

March 31, 2026March 31, 2026 king

VOS3000 Media Proxy and System Parameters: Complete Configuration Reference

VOS3000 media proxy and system parameters control the core functionality of your VoIP softswitch. Proper configuration of these parameters determines call quality, NAT traversal success, security levels, and overall system performance. This comprehensive reference guide covers all critical parameters from the official VOS3000 2.1.9.07 manual, explaining their functions and recommended configurations for different deployment scenarios.

📞 Need help configuring VOS3000 parameters? WhatsApp: +8801911119966

📡 Understanding Media Proxy in VOS3000

Media proxy determines whether RTP (Real-time Transport Protocol) voice packets flow directly between endpoints or through the VOS3000 server. This decision has significant implications for NAT traversal, audio quality, server resource usage, and call reliability.

📊 VOS3000 Media Proxy Modes

The SS_MEDIAPROXYMODE parameter controls media proxy behavior with four distinct modes:

Mode	Behavior	Server Load	Best Use Case
Off	Never proxy media; RTP flows directly between endpoints	Lowest	Public IP endpoints, no NAT issues
On	Always proxy all media through server	Highest	Troubleshooting, maximum control
Auto	Intelligent decision based on conditions	Variable	Mixed environments, recommended
Must On	Forced proxy regardless of other settings	Highest	Specific debugging scenarios only

⚙️ Media Proxy Auto Mode Decision Logic (VOS3000 Media Proxy)

When SS_MEDIAPROXYMODE is set to “Auto,” VOS3000 follows a precise decision algorithm to determine whether media proxy is needed:

Media Proxy Decision Steps (Auto Mode):

Step 1: Check if caller or callee MUST have media proxy
        ├── If gateway/phone has Media Proxy = Must On
        └── Result: ENABLE media proxy

Step 2: Check if caller or callee has Media Proxy disabled
        ├── If gateway/phone has Media Proxy = Off
        └── Result: DISABLE media proxy

Step 3: Check if caller or callee has Media Proxy enabled
        ├── If gateway/phone has Media Proxy = On
        └── Result: ENABLE media proxy

Step 4: Check if callee has local ring enabled
        ├── Local ring requires media proxy for ringback tone
        └── Result: ENABLE media proxy

Step 5: Check for dynamic registration with encryption
        ├── If phone/gateway uses dynamic register AND encryption
        └── Result: ENABLE media proxy

Step 6: Check cross-network routing (SS_MEDIAPROXYBETWEENNET)
        ├── If caller and callee from different networks
        └── Result: ENABLE media proxy

Step 7: Check NAT conditions (SS_MEDIAPROXYBEHINDNAT)
        ├── If phone and gateway in same NAT, SS_MEDIAPROXYSAMENAT = On
        ├── If phone and gateway in different NAT, one in private network
        └── Result: ENABLE media proxy

Step 8: Default action
        └── Result: DISABLE media proxy

🔧 Configuring Media Proxy Parameters

📍 Location in VOS3000 Client

Navigation Path:
Operation Management → Softswitch Management → Additional Settings → System Parameter

Parameter Name: SS_MEDIAPROXYMODE
Valid Values: Off, On, Auto, Must On
Default Value: Auto

Related Parameters:
┌─────────────────────────────────────────────────────────────┐
│ Parameter Name                  │ Description               │
├─────────────────────────────────────────────────────────────┤
│ SS_MEDIAPROXYBETWEENNET        │ Proxy for cross-network   │
│ SS_MEDIAPROXYBEHINDNAT         │ Proxy for behind-NAT      │
│ SS_MEDIAPROXYSAMENAT           │ Proxy for same-NAT        │
└─────────────────────────────────────────────────────────────┘

📡 RTP Port Configuration (VOS3000 Media Proxy)

RTP port configuration determines which UDP ports VOS3000 uses for voice media streams. Proper configuration is essential for firewall rules and capacity planning. VOS3000 Media Proxy

📊 RTP Port Parameters VOS3000 Media Proxy

Parameter	Default Value	Description
SS_RTP_PORT_RANGE	10000,39999	UDP port range for RTP media streams
SS_H245_PORT_RANGE	10000,39999	H.245 port range for H.323 calls
IVR_RTP_PORT	40000,47999	RTP port range for IVR services

⚙️ RTP Port Sizing Calculation

RTP Port Capacity Planning:

Each concurrent call uses 2 RTP ports (one for each direction)
Port Range: 10000-39999 = 30,000 ports
Maximum Concurrent Calls = 30,000 / 2 = 15,000 calls

However, consider:
- Each port allocation has overhead
- IVR services need separate port range
- H.323 calls share same range

Recommended Configuration by Capacity:
┌──────────────────────────────────────────────────────────────┐
│ Expected Capacity │ RTP Port Range    │ IVR Port Range      │
├──────────────────────────────────────────────────────────────┤
│ Small (<500 CC)   │ 10000-19999       │ 40000-40999         │
│ Medium (500-2000) │ 10000-29999       │ 40000-41999         │
│ Large (2000-5000) │ 10000-39999       │ 40000-44999         │
│ Enterprise (5000+)│ 10000-59999       │ 60000-64999         │
└──────────────────────────────────────────────────────────────┘

Firewall Rule Example:
iptables -A INPUT -p udp --dport 10000:39999 -j ACCEPT
iptables -A INPUT -p udp --dport 40000:47999 -j ACCEPT

🔑 SIP Parameters Reference – VOS3000 Media Proxy

SIP parameters control how VOS3000 handles SIP signaling, authentication, and session management. These parameters directly impact call setup success and session reliability.

📊 Critical SIP Parameters

Parameter	Default	Purpose
SS_SIP_NAT_KEEP_ALIVE_MESSAGE	HELLO	Content of NAT keep-alive message
SS_SIP_NAT_KEEP_ALIVE_PERIOD	30	Keep-alive interval in seconds (10-86400)
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL	500	Interval between sending keep-alives (ms)
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME	3000	Number of keep-alives sent per batch
SS_SIP_SESSION_TTL	1800	Session Timer TTL in seconds
SS_SIP_SESSION_UPDATE_SEGMENT	300	Session update interval in seconds
SS_SIP_RESEND_INTERVAL	0.5,1,2,4,4,4,4,4,4,4	SIP message resend intervals (seconds)
SS_SIP_NO_TIMER_REINVITE_INTERVAL	7200	Max call time for non-timer SIP clients

⚙️ NAT Keep-Alive Configuration

NAT Keep-Alive Purpose:
- Maintains NAT binding for devices behind NAT
- Prevents one-way audio caused by expired bindings
- Essential for devices that don't support SIP Timer

How It Works:
1. VOS3000 sends UDP message to registered device IP
2. Message content = SS_SIP_NAT_KEEP_ALIVE_MESSAGE (default: "HELLO")
3. Sent every SS_SIP_NAT_KEEP_ALIVE_PERIOD seconds (default: 30)
4. This keeps the NAT mapping active

Configuration Example:
SS_SIP_NAT_KEEP_ALIVE_MESSAGE = "HELLO"
SS_SIP_NAT_KEEP_ALIVE_PERIOD = 30
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL = 500
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME = 3000

This means:
- Send "HELLO" to each device every 30 seconds
- Wait 500ms between sending to different devices
- Process 3000 devices in each batch

Scaling Notes:
- 3000 devices × 500ms = 25 minutes to process all
- Adjust SEND_ONE_TIME for large deployments
- Increase SEND_INTERVAL if network is slow

🔐 Authentication Parameters

Authentication parameters control how VOS3000 handles SIP authentication challenges and account lockout policies for security.

📊 Authentication Security Parameters

Parameter	Default	Purpose
SS_AUTHENTICATION_MAX_RETRY	6	Max auth retries before suspension (0-999)
SS_AUTHENTICATION_FAILED_SUSPEND	180	Suspension duration in seconds (60-3600)
SS_SIP_AUTHENTICATION_CODE	Unauthorized(401)	SIP response code for auth challenge
SS_SIP_AUTHENTICATION_TIMEOUT	10	Timeout for SIP authentication in seconds
SS_SIP_AUTHENTICATION_RETRY	6	SIP auth retry count for 401/407 responses

⚙️ Authentication Lockout Configuration

Security Configuration Example:

For High-Security Environments:
SS_AUTHENTICATION_MAX_RETRY = 3
SS_AUTHENTICATION_FAILED_SUSPEND = 300

For Standard Environments:
SS_AUTHENTICATION_MAX_RETRY = 6
SS_AUTHENTICATION_FAILED_SUSPEND = 180

For Relaxed Environments (trusted networks only):
SS_AUTHENTICATION_MAX_RETRY = 10
SS_AUTHENTICATION_FAILED_SUSPEND = 60

How Lockout Works:
1. Device attempts registration with wrong password
2. VOS3000 returns 401 Unauthorized
3. Device retries (up to SS_AUTHENTICATION_MAX_RETRY times)
4. After max retries, IP is added to temporary block list
5. Block lasts for SS_AUTHENTICATION_FAILED_SUSPEND seconds
6. After timeout, device can retry

This protects against:
- Brute force password attacks
- SIP flood attacks
- Credential guessing
- Automated hacking tools

📊 Session Timer Configuration (VOS3000 Media Proxy)

Session timers ensure that hung calls are detected and cleaned up, preventing “ghost calls” and billing errors.

⚙️ Session Timer Parameters

Session Timer Configuration:

SS_SIP_SESSION_TTL = 1800 (30 minutes)
SS_SIP_SESSION_UPDATE_SEGMENT = 300 (5 minutes)
SS_SIP_NO_TIMER_REINVITE_INTERVAL = 7200 (2 hours)

How SIP Session Timer Works:
1. During call setup, session timer is negotiated
2. VOS3000 sends UPDATE or re-INVITE at interval
3. If no response, session is considered dead
4. Call is terminated and CDR is generated

For Non-Timer-Capable Clients:
- SS_SIP_NO_TIMER_REINVITE_INTERVAL sets max call time
- After this duration, call is terminated
- Prevents ultra-long "zombie" calls

Recommended Values:
┌────────────────────────────────────────────────────────────┐
│ Scenario           │ TTL  │ Update Segment │ Max No-Timer │
├────────────────────────────────────────────────────────────┤
│ Standard VoIP      │ 1800 │ 300            │ 7200         │
│ High-Volume Trunk  │ 3600 │ 600            │ 14400        │
│ Calling Card       │ 900  │ 180            │ 3600         │
│ Enterprise PBX     │ 1800 │ 300            │ 28800        │
└────────────────────────────────────────────────────────────┘

Session Timer Benefits:
- Detects hung calls automatically
- Prevents billing discrepancies
- Reduces "ghost call" complaints
- Frees system resources

🎯 H.323 Parameters Reference

For environments using H.323 protocol, VOS3000 provides comprehensive parameter controls.

📊 Critical H.323 Parameters

Parameter	Default	Purpose
SS_H245_PORT_RANGE	10000,39999	Port range for H.245 control channel
SS_H323_DTMF_METHOD	H.245 alphanumeric	Default DTMF transmission method
SS_H323_TIMEOUT_ALERTING	120	Timeout for alerting state (seconds)
SS_H323_TIMEOUT_CALLPROCEEDING	20	Timeout for call proceeding (seconds)
SS_H323_TIMEOUT_SETUP	5	Timeout for call setup (seconds)

📈 Quality of Service (QoS) Parameters

QoS parameters control the DSCP marking on IP packets for prioritization in managed networks.

⚙️ QoS Configuration

QoS Parameters:

SS_QOS_SIGNAL = 0xa0 (default)
- DSCP marking for SIP/H.323 signaling packets
- Hex value applied to IP header ToS field

SS_QOS_RTP = 0xa0 (default)
- DSCP marking for RTP media packets
- Hex value applied to IP header ToS field

DSCP Value Reference:
┌─────────────────────────────────────────────────────────────┐
│ Hex Value │ Binary  │ DSCP Class        │ Description      │
├─────────────────────────────────────────────────────────────┤
│ 0x00      │ 000000  │ Best Effort       │ Default, no QoS  │
│ 0x20      │ 001000  │ CS1               │ Scavenger        │
│ 0x40      │ 010000  │ CS2               │ OAM              │
│ 0x60      │ 011000  │ CS3               │ Signaling        │
│ 0x80      │ 100000  │ CS4               │ Real-time        │
│ 0xa0      │ 101000  │ CS5 / EF          │ Voice (default)  │
│ 0xc0      │ 110000  │ CS6               │ Network control  │
└─────────────────────────────────────────────────────────────┘

When to Configure:
- Only in managed networks with QoS policies
- Coordinate with network team on DSCP values
- Match router/switch QoS configuration

📊 Billing and CDR Parameters

These parameters control billing precision and CDR generation behavior. VOS3000 Media Proxy

⚙️ Critical Billing Parameters

Parameter	Default	Purpose
SERVER_BILLING_HOLD_TIME_PRECISION	50	Billing time precision in milliseconds
SERVER_MAX_CDR_PENDING_LIST_LENGTH	100000	Max pending CDR queue length
SERVER_CDR_FILE_WRITE_MAX	2048	Max CDR files to retain
SERVER_CDR_FILE_WRITE_INTERVAL	60	CDR file write interval (seconds)

📖 VOS3000 Extended Firewall Configuration
📖 VOS3000 One-Way Audio Troubleshooting
📖 VOS3000 Server Configuration Guide
📖 VOS3000 FAQ Based on Official Manual
📖 VOS3000 Downloads

❓ Frequently Asked Questions

Should I set media proxy to On or Auto?

Auto mode is recommended for most deployments. It intelligently enables media proxy only when needed (NAT traversal, encryption, cross-network calls) while allowing direct RTP when possible. This provides the best balance of reliability and server resource usage.

How do I know if my RTP port range is sufficient?

Calculate: Each concurrent call uses 2 RTP ports. With default range 10000-39999 (30,000 ports), you can support 15,000 concurrent calls. Monitor port usage through system performance monitoring. If you see port allocation errors, increase the range or reduce concurrent call load.

Why do calls drop at 30 seconds?

This typically indicates SIP session timer or NAT binding issues. Check SS_SIP_SESSION_TTL and ensure NAT keep-alive is configured. The 30-second timeout often corresponds to NAT binding expiry when keep-alives are not working.

What is the best authentication retry setting?

For most environments, the default of 6 retries with 180-second suspension works well. For high-security environments, reduce to 3 retries with longer suspension (300+ seconds). Balance security against false positives from legitimate users mistyping passwords.

How do I troubleshoot media proxy issues?

Use Debug Trace in VOS3000 to capture SIP and SDP messages. Check if media proxy is being invoked (look at the c= line in SDP). Verify that RTP ports are within configured range. Check firewall rules allow both signaling and RTP ports.

📞 Get Expert Help with VOS3000 Configuration

Need assistance optimizing VOS3000 parameters for your specific deployment? Our team provides professional VOS3000 installation, configuration, and performance tuning services.

📱 WhatsApp: +8801911119966

📞 Need Professional VOS3000 Setup Support?

For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:

📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads

VOS3000 softswitch VoIP, VOS3000 seguridad, VOS3000 Call Center Soluciones, VOS3000 API Integración, VOS3000 Infraestructura, VOS3000 Errores Ruting Llamadas, VOS3000错误代码替换与呼叫失败排查, VOS3000 Optimización de Rendimiento, VOS3000 Códigos Error Terminación, VOS3000 NoAvailableRouter错误解决方案, Negocio VoIP Mayorista, VICIDIAL Servidor, Softswitch Barato, VoIP批发业务, 软交换比较, Advance Routing, VOS3000 Troubleshooting Guide, VOS3000 CDR Analysis, Guía Completa VOS3000 2026, VOS3000 指南 2026, SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons

VOS3000 System Parameters & Timers: Important Guide

March 8, 2026March 8, 2026 king

VOS3000 System Parameters & Timers: Important Guide

VOS3000 contains hundreds of configurable parameters that control every aspect of its operation – from SIP timers and H.323 settings to billing rules and alarm thresholds. Understanding these VOS3000 system parameters is essential for tuning performance, troubleshooting issues, and customizing the platform to your specific needs.

This comprehensive reference covers the most important parameters grouped by category, with explanations of what they do and when you might need to change them.

Where to Find VOS3000 System Parameters

VOS3000 parameters are spread across two main locations:

System Management > System Parameter – server‑level parameters (database, reports, passwords, etc.)
Operation Management > Softswitch Management > Additional Settings > System Parameter – softswitch runtime parameters (SIP, H.323, media, routing)

Changes to parameters take effect immediately – no service restart required in most cases.

VOS3000 Server Parameters (System Management)

These parameters control the VOS3000 server environment, database behavior, and reporting.

Parameter Name	Default Value	Description	When to Change
`SERVER_BILLING_FEE_PRECISION`	0.0000000	Number of decimal places for billing amounts.	If you need more/less precision in call charges (e.g., 4 decimals for fractional cents).
`SERVER_BILLING_HOLD_TIME_PRECISION`	1000	Time rounding precision in milliseconds. E.g., 50 means round to nearest 50ms.	Adjust to match your carrier’s billing increments (6 seconds = 6000).
`SERVER_QUERY_ONE_PAGE_SIZE`	10000	Number of records displayed per page in CDR queries.	Increase if you want to see more records at once (may slow down browser).
`SERVER_QUERY_CDR_MAX_DAY_INTERVAL`	31	Maximum number of days allowed in a single CDR query.	Increase for longer reports, but beware of performance impact.
`SERVER_ALARM_EMAIL`	(empty)	Email address for alarm notifications.	Set to receive email alerts when alarms trigger.
`SERVER_ALARM_ENABLE_EMAIL`	Off	Enable/disable email alarms.	Turn On after configuring email settings.
`SERVER_PASSWORD_LENGTH`	8	Minimum password length for new users.	Increase for better security (e.g., 12).
`SERVER_PAY_DELAY_CUSTOMER_EXPIRE_DAY`	365	Days added to account expiry after recharge.	Adjust based on your recharge policies.
`SERVER_REPORT_*`	Various	Enable/disable automatic generation of daily reports.	Turn off reports you don’t need to save server resources.

Softswitch SIP Parameters (VOS3000 System Parameters)

These parameters control SIP signaling behavior and are critical for interoperability with carriers and devices.

Parameter Name	Default Value	Description	When to Change
`SS_SIP_TIMEOUT_INVITE`	10	Seconds to wait for a response to INVITE before trying next gateway.	Increase if carriers are slow to respond; decrease to fail faster.
`SS_SIP_TIMEOUT_RINGING`	120	Seconds to wait for answer after receiving ringing (180).	Adjust for markets where users take longer to answer.
`SS_SIP_TIMEOUT_TRYING`	20	Seconds to wait for 100 Trying after INVITE.	Increase if carriers don’t send early progress.
`SS_SIP_TIMEOUT_SESSION_PROGRESS`	20	Seconds to wait for 183 Session Progress.	Some carriers send 183 very late – increase if calls fail prematurely.
`SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP`	120	Seconds to wait for 183 with SDP (early media).	Increase if early media takes time to arrive.
`SS_SIP_STOP_SWITCH_AFTER_SDP`	On	Stop trying other gateways after receiving SDP (media negotiation started).	Turn Off if you want to continue trying better gateways even after SDP received.
`SS_SIP_RESEND_INTERVAL`	0.5,1,2,4,4,4,4,4,4,4	Comma‑separated retransmission intervals (seconds) for SIP messages.	Customize for networks with high packet loss (longer intervals).
`SS_SIP_SESSION_TTL`	600	Session timer interval (seconds) for keeping calls alive.	Shorter for aggressive dead‑call detection; longer to reduce signaling.
`SS_SIP_NO_TIMER_REINVITE_INTERVAL`	7200	Maximum call duration (seconds) for non‑timer‑aware SIP devices.	Force hangup of very long calls to prevent billing errors.
`SS_SIP_NAT_KEEP_ALIVE_PERIOD`	30	Seconds between NAT keep‑alive messages.	Reduce if devices behind NAT drop pinholes quickly.

Softswitch H.323 Parameters (VOS3000 System Parameters)

For networks using H.323 gateways or terminals.

Parameter Name	Default Value	Description	When to Change
`SS_H323_TIMEOUT_SETUP`	5	Seconds to wait for Call Proceeding after Setup.	Increase if H.323 gateways are slow.
`SS_H323_TIMEOUT_CALLPROCEEDING`	20	Seconds to wait for Alerting after Call Proceeding.	Adjust based on typical answer times.
`SS_H323_TIMEOUT_ALERTING`	120	Seconds to wait for Connect after Alerting.	Same as SIP ringing timeout.
`SS_H323_TIMEOUT_CALLPROCEEDING_OLC`	20	Seconds to wait for OLC (Open Logical Channel) after Call Proceeding.	Increase if media negotiation is slow.
`SS_H323_STOP_SWITCH_AFTER_OLC`	Off	Stop trying other gateways after OLC (media opened).	Turn On if you want to lock the gateway once media starts.

System‑Wide Softswitch Parameters

These affect overall call handling and routing logic.

Parameter Name	Default Value	Description	When to Change
`SS_MAX_CALL_DURATION`	None	Global maximum call length in seconds.	Set to prevent extremely long calls (e.g., 10800 for 3 hours).
`SS_MEDIA_PROXY_MODE`	Auto	Media proxy decision: Auto, On, Off, Must On.	Force On if you need recording or NAT traversal for all calls.
`SS_MEDIA_PROXY_PORT_RANGE`	10000,39999	RTP port range for media proxy.	Adjust if you need to limit firewall rules.
`SS_GATEWAY_ASR_CALCULATE`	Off	Enable real‑time ASR (Answer Seizure Ratio) calculation for routing.	Turn On to use ASR as a routing metric.
`SS_GATEWAY_ACD_CALCULATE`	Off	Enable real‑time ACD (Average Call Duration) calculation.	Turn On to use ACD in routing decisions.
`SS_GATEWAY_SWITCH_UNTIL_CONNECT`	Off	Continue trying gateways until one answers (not just until ringing).	Useful when carriers rarely answer but you want to try all options.
`SS_REDIRECT_OFFLINE_PHONE_TO_GATEWAY`	Off	If a called phone is offline, try routing through gateways.	Useful for hybrid networks where phones may not always register.
`SS_ACCOUNT_INDICATION_METHOD`	Off	How to warn of low balance: Off, Prompt balance, Prompt duration.	Enable to play warnings to callers before cutoff.

Audio Service (IVR) Parameters

Controls for IVR, callback, and value‑added services.

Parameter Name	Default Value	Description	When to Change
`IVR_RINGING_TIMEOUT`	120	Seconds to wait for answer in IVR scenarios.	Adjust for different user behavior.
`IVR_SETUP_TIMEOUT`	20	Seconds to wait for initial response.	Increase if IVR connections are slow.
`IVR_MEDIA_CHECK_TIMEOUT`	2	Minutes of no media before hanging up.	Reduce to free ports faster on dead calls.
`IVR_CODEC_PRIORITY`	g729a,g729,g723,g711a,g711u	Preferred codec order for IVR.	Reorder based on your termination costs/quality.

Best Practices for Parameter Tuning – VOS3000 System Parameters

Change one parameter at a time and observe the effect.
Document your changes – keep a record of what you changed and why.
Test in a non‑production environment first if possible.
Be conservative with timeouts – too short causes failures, too long wastes resources.
Monitor call logs after changes to detect unintended consequences.

Frequently Asked Questions (VOS3000 System Parameters)

Do I need to restart VOS3000 after changing parameters?

No. VOS3000 reads parameters from the database in real time. Changes take effect immediately for new calls. Ongoing calls continue with the parameters they started with.

Can I break my system by changing a parameter?

Most parameters are safe to experiment with, but extreme values (e.g., setting timeouts to 0) can cause unexpected behavior. Always note the original value so you can revert if needed.

What’s the most important parameter for reducing call failures?

For SIP, start with SS_SIP_TIMEOUT_INVITE and SS_SIP_RESEND_INTERVAL. If carriers are slow to respond, increasing these can reduce “Response timeout” failures.

How do I enable NAT keep‑alive for SIP devices?

Set SS_SIP_NAT_KEEP_ALIVE_PERIOD to 20‑30 seconds and SS_SIP_NAT_KEEP_ALIVE_MESSAGE to “HELLO” or any string. The softswitch will send UDP packets to keep NAT bindings open.

What does “SS_MEDIAPROXYMODE = Auto” actually do?

Auto mode enables media proxy only when needed – e.g., when devices are behind different NATs, when encryption is required, or when a device explicitly requests it. This is the recommended setting for most deployments.

Conclusion

Mastering VOS3000 system parameters gives you fine‑grained control over your softswitch. Use this reference as a starting point, experiment carefully, and always monitor the impact of your changes. With the right tuning, you can maximize call completion rates, improve voice quality, and optimize resource usage.

Need expert help with VOS3000 configuration or performance tuning? Contact us on WhatsApp: +8801911119966

Further Resources

VOS3000 Installation, VOS3000 Server, VOS3000 SoftSwitch, VOS3000 Switch, VOS3000, VOS3000 Pricem VOS3000 Web, VOS3000 API, VOS3000 Rent, VOS3000 Manual, VOS3000 Downloads, VOS3000 VoIP, VOS3000 Carrier Switch, VOS3000, VOS3000 Login, VOS3000 Monitoring, VOS3000 Performance Metrics, VOS3000 Call Routing, VOS3000 Security, VOS3000 Web Manager, VOS3000 Versions, VOS3000 BillingVOS3000 Monitoring,VOS3000 Capacity, VOS3000 Billing System, VOS3000 License,VOS3000 Installation, VOS3000 Server, VOS3000 SoftSwitch, VOS3000 Switch, VOS3000, VOS3000 Pricem VOS3000 Web, VOS3000 API, VOS3000 Rent, VOS3000 Manual, VOS3000 Downloads, VOS3000 VoIP, VOS3000 Carrier Switch, VOS3000, VOS3000 Login, VOS3000 Monitoring, VOS3000 Performance Metrics, VOS3000 Call Routing, VOS3000 Security, VOS3000 Web Manager, VOS3000 Versions, VOS3000 BillingVOS3000 Monitoring,VOS3000 Capacity, VOS3000 Billing System, VOS3000 License, Mobile Apps for VOS3000, VOS3000 Mobile Apps, Mobile Apps, VOS3000 Apps, Android VOS3000, VOS3000 in IOS, Manual for VOS3000, VOS3000 Manual, Manual VOS3000, Reference Manual VOS3000, User Manual VOS3000, VOS安装, VOS3000 Security, VOS3000 托管, VOS3000 architecture, VOS3000 call termination, VOS3000 Data Maintenance, VOS3000 Disaster Recovery, VOS3000 System Parameters

VOS3000 Server Rent – Dedicated Best VoIP Servers with Pre-Installed VOS3000 2.1.9.07 for High Concurrent Calls

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

Table of Contents

🔍 What is VOS3000 Session Timer?

📊 The Zombie Call Problem

⚙️ VOS3000 Session Timer Parameters Complete Reference

📊 Core Session Timer Parameters Table

📊 Session Timer Refresh Calculation

📐 Session Timer Refresh Interval Formula

📡 NAT Keep-Alive Configuration Deep Dive

📊 NAT Keep-Alive Parameters Table

🔄 How NAT Keep-Alive Works in VOS3000

🔧 VOS3000 Session Timer Configuration Guide

📍 Navigation to System Parameters

⚙️ Recommended Configuration by Deployment Type

📊 Session Timer Message Flow Diagram

🚨 Session Timer Troubleshooting Guide

📊 Common Problems and Solutions

🔧 Debug Trace Analysis for Session Timer

📊 Session Timer vs NAT Keep-Alive Comparison

💰 VOS3000 Installation and Support Services

🔗 Related Resources

❓ Frequently Asked Questions about VOS3000 Session Timer

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

Why are my calls dropping exactly at 30 seconds?

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

How do I calculate the optimal session timer refresh interval?

Can session timer prevent billing fraud?

📞 Get Expert VOS3000 Session Timer Support

📞 Need Professional VOS3000 Setup Support?

VOS3000 Media Proxy and System Parameters: Complete Configuration Reference

Table of Contents

📡 Understanding Media Proxy in VOS3000

📊 VOS3000 Media Proxy Modes

⚙️ Media Proxy Auto Mode Decision Logic (VOS3000 Media Proxy)

🔧 Configuring Media Proxy Parameters

📍 Location in VOS3000 Client

📡 RTP Port Configuration (VOS3000 Media Proxy)

📊 RTP Port Parameters VOS3000 Media Proxy

⚙️ RTP Port Sizing Calculation

🔑 SIP Parameters Reference – VOS3000 Media Proxy

📊 Critical SIP Parameters

⚙️ NAT Keep-Alive Configuration

🔐 Authentication Parameters

📊 Authentication Security Parameters

⚙️ Authentication Lockout Configuration

📊 Session Timer Configuration (VOS3000 Media Proxy)

⚙️ Session Timer Parameters

🎯 H.323 Parameters Reference

📊 Critical H.323 Parameters

📈 Quality of Service (QoS) Parameters

⚙️ QoS Configuration

📊 Billing and CDR Parameters

⚙️ Critical Billing Parameters

🔗 Related Resources – VOS3000 Media Proxy

❓ Frequently Asked Questions

Should I set media proxy to On or Auto?

How do I know if my RTP port range is sufficient?

Why do calls drop at 30 seconds?

What is the best authentication retry setting?

How do I troubleshoot media proxy issues?

📞 Get Expert Help with VOS3000 Configuration

📞 Need Professional VOS3000 Setup Support?

VOS3000 System Parameters & Timers: Important Guide

Table of Contents

Where to Find VOS3000 System Parameters

VOS3000 Server Parameters (System Management)

Softswitch SIP Parameters (VOS3000 System Parameters)

Softswitch H.323 Parameters (VOS3000 System Parameters)

System‑Wide Softswitch Parameters

Audio Service (IVR) Parameters

Best Practices for Parameter Tuning – VOS3000 System Parameters

Frequently Asked Questions (VOS3000 System Parameters)

Do I need to restart VOS3000 after changing parameters?

Can I break my system by changing a parameter?

What’s the most important parameter for reducing call failures?

How do I enable NAT keep‑alive for SIP devices?

What does “SS_MEDIAPROXYMODE = Auto” actually do?

Conclusion

Further Resources