VOS3000 SIP No Timer Call Duration: Important Maximum Limit Guide

📞 Have you ever discovered runaway calls in your CDR records — sessions lasting hours beyond the actual conversation? The VOS3000 SIP no timer call duration parameter is your ultimate safety net. When SIP endpoints do not support session timers, this critical setting enforces a hard maximum limit, preventing zombie calls from draining your VoIP revenue. ⏱️

🚨 Not every SIP device implements RFC 4028 session timers. Legacy gateways, softphones, and some SIP trunks simply never include a Session-Expires header in their INVITE messages. For these non-timer endpoints, VOS3000 cannot actively verify if the call is still alive — and without a hard cap, orphaned calls can run indefinitely, generating phantom charges. The SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter solves this by imposing a maximum conversation time that VOS3000 enforces automatically. 🔐

🎯 This guide covers everything about the VOS3000 SIP no timer call duration — from the official default of 7200 seconds (2 hours) to recommended values by deployment type, its relationship with session timers, and step-by-step configuration to protect your billing accuracy.

🔐 What Is VOS3000 SIP No Timer Call Duration?

⏰ The VOS3000 SIP no timer call duration is controlled by the parameter SS_SIP_NO_TIMER_REINVITE_INTERVAL. It defines the maximum allowed conversation time for SIP callers that do NOT support the “timer” feature as defined in RFC 4028.

💡 Why this matters: When a SIP caller supports session timers, VOS3000 can periodically send re-INVITE or UPDATE messages to confirm the call is still connected. But when the caller does not support timers:

• ❌ No re-INVITE or UPDATE messages can be sent to verify the session
• ❌ VOS3000 cannot detect whether the far end is still alive
• ⚠️ The only protection is a hard timeout — once exceeded, the call is forcibly terminated
• 🛡️ Without this parameter, zombie calls could persist indefinitely

📍 Location in VOS3000 Client: Navigation → Operation management → Softswitch management → Additional settings → SIP parameter

📋 Official Parameter Specification

🔧 According to the VOS3000 2.1.9.07 Official Manual (Table 4-3, Section 4.3.5.2):

Attribute	Value
📌 Parameter Name	SS_SIP_NO_TIMER_REINVITE_INTERVAL
🔢 Default Value	7200
📐 Unit	Seconds
📝 Description	Maximum Conversation Time for Non-TIMER SIP Caller. If SIP caller doesn’t support “timer”, softswitch will stop the call when the time is up.

⏱️ Default of 7200 seconds = 2 hours. This means that by default, a call from a non-timer SIP endpoint will be forcibly terminated after 2 hours of continuous conversation — regardless of whether the call is still active or has become a zombie.

🔄 VOS3000 SIP No Timer Call Duration vs. Session Timer

📊 Understanding the relationship between the VOS3000 SIP no timer call duration and the session timer is essential for proper configuration. These two mechanisms work as complementary systems:

Aspect	Session Timer (RFC 4028)	No Timer Call Duration
📌 Parameter	SS_SIP_SESSION_TTL	SS_SIP_NO_TIMER_REINVITE_INTERVAL
🔢 Default	600s (10 min)	7200s (2 hours)
🎯 Applies When	Caller supports “timer”	Caller does NOT support “timer”
📡 Detection Method	Active — sends re-INVITE/UPDATE	Passive — hard timeout only
🔍 Session-Expires Header	Present in SIP messages	Not present
📞 Verification	Periodic refresh with 200 OK	None — just countdown
❌ Call Termination	No 200 OK → BYE sent	Time exceeded → BYE sent
🛡️ Protection Level	High — active probing	Lower — passive timeout

💡 Key takeaway: The VOS3000 session timer provides active call verification for timer-capable endpoints. The VOS3000 SIP no timer call duration provides passive protection for endpoints that lack timer support. Both are essential for a complete call management strategy.

🎯 How VOS3000 Decides Which Mechanism to Use

🖥️ When a SIP INVITE arrives at VOS3000, the softswitch inspects the SIP headers to determine whether the caller supports session timers:

📞 SIP INVITE Arrives at VOS3000
    │
    ├── VOS3000 checks for Session-Expires header
    │
    ├── ✅ Session-Expires header FOUND
    │   ├── Caller supports RFC 4028 session timer
    │   ├── VOS3000 uses SS_SIP_SESSION_TTL (default: 600s)
    │   ├── Active probing with re-INVITE/UPDATE messages
    │   └── Call verified every TTL/Segment interval
    │
    └── ❌ Session-Expires header NOT FOUND
        ├── Caller does NOT support session timer
        ├── VOS3000 uses SS_SIP_NO_TIMER_REINVITE_INTERVAL (default: 7200s)
        ├── NO active probing — passive countdown only
        └── Call forcibly terminated when time exceeds limit

⚙️ SS_SIP_NO_TIMER_REINVITE_INTERVAL — Deep Dive

🔐 Let’s examine the VOS3000 SIP no timer call duration parameter in full detail — what it does, how it works, and what happens when the limit is reached.

🔑 How the Parameter Works

⏱️ When a SIP caller that does not support session timers establishes a call through VOS3000:

1. 📞 The call is established normally (INVITE → 200 OK → ACK)
2. 🖥️ VOS3000 detects the absence of a Session-Expires header
3. ⏰ VOS3000 starts a countdown timer set to SS_SIP_NO_TIMER_REINVITE_INTERVAL seconds
4. 📊 The call proceeds normally while the countdown runs
5. 🚨 When the countdown reaches zero, VOS3000 sends a BYE message to terminate the call

⚠️ Important: Unlike session timers, VOS3000 does NOT send any re-INVITE or UPDATE messages during the call. The only action taken is the forced termination when the timer expires. This is a passive safety mechanism — it cannot detect whether the call is still alive before the timeout.

📊 Duration Conversion Table

📋 Common SS_SIP_NO_TIMER_REINVITE_INTERVAL values and their equivalent durations:

Seconds	Minutes	Hours	Common Name
900	15	0.25	Quarter hour
1800	30	0.5	Half hour
3600	60	1	One hour
5400	90	1.5	Ninety minutes
7200	120	2	✅ Default (two hours)
10800	180	3	Three hours
14400	240	4	Four hours

🛡️ Preventing Runaway Calls with VOS3000 SIP No Timer Call Duration

🚨 Runaway calls are one of the most costly problems in VoIP operations. They occur when a call remains in “connected” state long after both parties have stopped talking — typically because of network failures, endpoint crashes, or NAT timeouts that prevent proper BYE messages.

⚠️ How Runaway Calls Happen

📞 Here’s the scenario that creates runaway calls on non-timer endpoints:

📞 Call Established Between Non-Timer Endpoint and VOS3000
    │
    ├── Both parties talk normally
    │
    ├── 🔴 Network failure / endpoint crash / NAT timeout
    │   ├── No BYE message sent (endpoint is dead/unreachable)
    │   ├── Call remains in "connected" state on VOS3000
    │   └── VOS3000 CANNOT send re-INVITE (endpoint has no timer support)
    │
    ├── ⏰ Without SS_SIP_NO_TIMER_REINVITE_INTERVAL:
    │   └── ❌ Call stays connected INDEFINITELY
    │       └── 💸 Billing continues to accumulate
    │
    └── ✅ With SS_SIP_NO_TIMER_REINVITE_INTERVAL = 7200s:
        └── After 2 hours, VOS3000 sends BYE
            └── 🛡️ Call terminated, billing stops

💡 Critical point: Unlike timer-capable endpoints where VOS3000 can actively probe the session, non-timer endpoints offer zero visibility into call health. The SS_SIP_NO_TIMER_REINVITE_INTERVAL is the only mechanism that prevents indefinite zombie calls.

📊 Runaway Call Cost Impact Table

💸 Understanding the financial impact of runaway calls shows why the VOS3000 SIP no timer call duration setting matters:

Zombie Call Duration	Rate ($/min)	Cost per Incident	10 Incidents/Month
1 hour (no limit)	$0.02	$1.20	$12.00
4 hours (no limit)	$0.02	$4.80	$48.00
12 hours (no limit)	$0.02	$14.40	$144.00
24 hours (no limit)	$0.05	$72.00	$720.00
48 hours (no limit)	$0.10	$288.00	$2,880.00

🚨 As you can see, without a hard call duration limit, a single zombie call on a premium route can cost hundreds of dollars. The VOS3000 SIP no timer call duration parameter ensures that even if the endpoint cannot be actively probed, the call will be terminated within a predictable timeframe.

📊 VOS3000 SIP No Timer Call Duration and Billing Accuracy

💰 Billing accuracy is directly affected by the VOS3000 SIP no timer call duration setting. Here’s how:

🔐 Billing Impact Analysis

NO_TIMER_INTERVAL	Max Zombie Duration	Billing Risk	CDR Accuracy
900s (15 min)	15 minutes max	🛡️ Very Low	✅ Excellent
1800s (30 min)	30 minutes max	✅ Low	✅ Very Good
3600s (1 hour)	1 hour max	🔧 Medium-Low	📊 Good
7200s (2 hours) ✅	2 hours max	⚠️ Medium	📊 Acceptable
14400s (4 hours)	4 hours max	🚨 High	❌ Poor
Not configured	Unlimited	🔥 Critical	❌ Very Poor

📝 Billing accuracy depends on CDR records matching actual call durations. When zombie calls persist, CDRs show inflated durations that do not correspond to real conversations. This creates CDR billing discrepancies that can erode customer trust and cause revenue disputes. For more on the overall billing framework, see our VOS3000 billing system guide.

🔧 Step-by-Step Configuration of VOS3000 SIP No Timer Call Duration

🖥️ Follow these steps to configure SS_SIP_NO_TIMER_REINVITE_INTERVAL in your VOS3000 softswitch:

Step 1: Navigate to SIP Parameters 📋

1. 🔐 Log in to VOS3000 Client with administrator credentials
2. 📌 Navigate: Operation management → Softswitch management → Additional settings → SIP parameter
3. 🔍 Locate SS_SIP_NO_TIMER_REINVITE_INTERVAL in the SIP parameter list

Step 2: Choose Your Value ⏱️

🎯 Select the appropriate value based on your deployment type:

Deployment Type	Recommended Value	Duration	Rationale
🏢 Standard enterprise	7200s	2 hours	✅ Default — sufficient for most calls
📞 Wholesale termination	3600s	1 hour	🔧 Tighter control, lower risk
🛡️ Premium / high-value routes	1800s	30 minutes	🔐 Maximum billing protection
🌐 Legacy gateway networks	1800s–3600s	30–60 min	📡 Old devices often lack timer support
📞 Call center operations	5400s	90 minutes	📊 Accommodates long agent calls
🔥 Maximum protection	900s	15 minutes	🛡️ Zero tolerance for runaway calls

Step 3: Apply and Save ✅

1. 📝 Enter the desired value (in seconds) in the SS_SIP_NO_TIMER_REINVITE_INTERVAL field
2. 💾 Click Save to apply the configuration
3. 🔄 The new value takes effect for all subsequent calls from non-timer SIP endpoints

⚠️ Note: Existing calls are not affected by the change. Only new calls established after the configuration update will use the new interval value.

🔄 Relationship with Other VOS3000 Parameters

🔗 The VOS3000 SIP no timer call duration does not operate in isolation. It works alongside several related parameters that together form a comprehensive call management system:

Parameter	Default	Unit	Relationship to NO_TIMER
SS_SIP_SESSION_TTL	600	Seconds	🔄 Complementary — applies when timer IS supported
SS_SIP_SESSION_UPDATE_SEGMENT	2	Count	📊 Controls re-INVITE frequency for timer calls
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP	0	Seconds	⏰ Grace period — applies only to timer calls
SS_MAX_CALL_DURATION	None	—	🛡️ System-level hard limit for ALL calls

💡 Key relationship: The SS_MAX_CALL_DURATION parameter (system parameter, not SIP parameter) enforces a hard maximum call duration for all calls regardless of whether they support timers or not. If both SS_SIP_NO_TIMER_REINVITE_INTERVAL and SS_MAX_CALL_DURATION are configured, the shorter of the two values takes effect. Read more about this in our VOS3000 max call duration guide and system parameters overview.

📋 Parameter Interaction Flow

📞 Call Arrives at VOS3000
    │
    ├── Check: Does SS_MAX_CALL_DURATION exist?
    │   ├── YES → Apply system-level hard limit
    │   └── NO  → No system-level limit
    │
    ├── Check: Does caller support "timer"?
    │   ├── YES → Apply SS_SIP_SESSION_TTL (600s default)
    │   │        Active probing via re-INVITE/UPDATE
    │   │        Hang up if no 200 OK confirmation
    │   │
    │   └── NO  → Apply SS_SIP_NO_TIMER_REINVITE_INTERVAL (7200s default)
    │            NO active probing — passive countdown
    │            Hang up when time exceeded
    │
    └── 🛡️ Effective limit = min(SS_MAX_CALL_DURATION, applicable timer)

💡 Best Practices for VOS3000 SIP No Timer Call Duration

🎯 Follow these best practices to maximize the effectiveness of your VOS3000 SIP no timer call duration configuration:

Best Practice	Recommendation	Reason
🎯 Set SS_MAX_CALL_DURATION	Configure a system-level limit as backup	🛡️ Double protection for all calls
📊 Monitor CDR records	Check for calls near the 7200s limit weekly	🔍 Detects non-timer endpoint patterns
📞 Encourage timer support	Ask vendors to enable RFC 4028 on endpoints	✅ Active probing is far superior
🔧 Lower for premium routes	Set 1800s–3600s for expensive destinations	🔐 Minimizes billing exposure
🔄 Coordinate with session timer	NO_TIMER should be ≥ 3× SS_SIP_SESSION_TTL	📊 Consistent protection across both modes
📝 Document configuration	Record all timer-related parameter values	📋 Simplifies troubleshooting later
📡 Verify endpoint compatibility	Capture SIP INVITE to check Session-Expires	🔍 Confirms which mode is active

💡 Pro tip: If most of your SIP trunks support session timers, a higher VOS3000 SIP no timer call duration (7200s default) is acceptable since only a few calls will hit this limit. But if you have many legacy gateways without timer support, lower the value to 1800s–3600s for better protection. Check our VOS3000 parameter description guide for the complete parameter reference.

🛡️ Common Problems and Troubleshooting

⚠️ Here are the most common issues related to the VOS3000 SIP no timer call duration and their solutions:

❌ Problem 1: Calls Being Cut After Exactly 2 Hours

🔍 Symptom: Legitimate long-duration calls are being terminated at exactly 2 hours.

💡 Cause: The SIP caller does not support session timers, and SS_SIP_NO_TIMER_REINVITE_INTERVAL is set to the default 7200 seconds.

✅ Solutions:

• 🔧 Increase SS_SIP_NO_TIMER_REINVITE_INTERVAL if 2-hour calls are expected
• 📞 Ask the SIP endpoint vendor to implement RFC 4028 session timer support
• 🔐 Verify the call flow using our SIP call flow guide

❌ Problem 2: Ultra-Long Bills from Non-Timer Endpoints

🔍 Symptom: CDR records show calls lasting the full 7200 seconds, but the actual conversation was much shorter.

💡 Cause: The endpoint crashed or lost network connectivity without sending BYE, and the non-timer interval is too long.

✅ Solutions:

• ⏱️ Reduce SS_SIP_NO_TIMER_REINVITE_INTERVAL to 1800s or 3600s
• 🛡️ Set SS_MAX_CALL_DURATION as a secondary safety limit
• 📊 Cross-reference CDR records with billing system data

❌ Problem 3: Not Sure Which Endpoints Support Session Timers

🔍 Symptom: Unknown whether your SIP trunks and gateways support RFC 4028.

💡 Solution: Capture the SIP INVITE message and check for the Session-Expires header:

# SIP INVITE from a TIMER-capable endpoint:
INVITE sip:destination@example.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060
Session-Expires: 600           <-- ✅ Timer SUPPORTED
Min-SE: 90
...

# SIP INVITE from a NON-TIMER endpoint:
INVITE sip:destination@example.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060
                                <-- ❌ No Session-Expires header
...
# VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL for this call

📞 Need more help with SIP debugging? See our VOS3000 troubleshooting guide for detailed instructions.

📊 Complete VOS3000 SIP No Timer Call Duration Decision Matrix

🎯 Use this decision matrix to select the optimal SS_SIP_NO_TIMER_REINVITE_INTERVAL value for your deployment:

Factor	Low Value (900–1800s)	Mid Value (3600–5400s)	High Value (7200s+)
🛡️ Billing risk	✅ Very low	🔧 Moderate	⚠️ Higher
📞 Call disruption	⚠️ Possible for long calls	✅ Rare	✅ Very rare
💸 Zombie call cost	✅ Minimal	🔧 Controlled	⚠️ Potentially high
📊 CDR accuracy	✅ Excellent	📊 Good	🔧 Acceptable
🎯 Best for	Premium routes, high rates	Wholesale, mixed traffic	Standard enterprise, low rates

❓ Frequently Asked Questions

❓ What is the default VOS3000 SIP no timer call duration?

⏱️ The default VOS3000 SIP no timer call duration is 7200 seconds (2 hours), configured via the SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter. This means that when a SIP caller does not support the “timer” feature, VOS3000 will forcibly terminate the call after 7200 seconds of continuous conversation. This default is defined in the VOS3000 2.1.9.07 Official Manual (Table 4-3, Section 4.3.5.2).

❓ What happens when VOS3000 SIP no timer call duration is exceeded?

🚨 When the call duration from a non-timer SIP endpoint exceeds the SS_SIP_NO_TIMER_REINVITE_INTERVAL value, VOS3000 sends a BYE message to terminate the call on both legs. The call is removed from the active call list, and a CDR record is generated with the total duration. This is a hard termination — there is no grace period or retry mechanism for non-timer calls.

❓ How is VOS3000 SIP no timer call duration different from session timer?

🔄 The key difference is the detection method. The VOS3000 session timer (SS_SIP_SESSION_TTL, default 600s) actively probes timer-capable endpoints using re-INVITE/UPDATE messages. The VOS3000 SIP no timer call duration (SS_SIP_NO_TIMER_REINVITE_INTERVAL, default 7200s) is a passive countdown — no probing occurs, and the call is simply terminated when the time limit is reached. Session timer is for endpoints that support RFC 4028; the no timer interval is for endpoints that do not.

❓ Can I set SS_SIP_NO_TIMER_REINVITE_INTERVAL to unlimited?

❌ While technically possible, setting the VOS3000 SIP no timer call duration to an extremely high value (or leaving it unconfigured) is strongly discouraged. Without a limit, zombie calls from non-timer endpoints can persist indefinitely, generating phantom billing charges. Always set a reasonable value based on your expected maximum call duration and risk tolerance. Also configure SS_MAX_CALL_DURATION as a secondary safety mechanism.

❓ Does VOS3000 SIP no timer call duration affect calls that support session timers?

📱 No. The SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter only applies when the SIP caller does NOT support the “timer” feature. If the caller includes a Session-Expires header in the INVITE or 200 OK messages, VOS3000 uses the session timer mechanism (SS_SIP_SESSION_TTL) instead. The two mechanisms are mutually exclusive — each call uses one or the other based on the endpoint’s timer support.

❓ How do I check if my SIP endpoints support session timers?

🔍 Capture the SIP INVITE message using a network analyzer like Wireshark or the VOS3000 built-in SIP trace. Look for the Session-Expires header in the INVITE message. If the header is present, the endpoint supports RFC 4028 session timers and VOS3000 will use SS_SIP_SESSION_TTL. If the header is absent, the endpoint does not support timers and VOS3000 will use the VOS3000 SIP no timer call duration instead. See our troubleshooting guide for detailed SIP trace instructions.

❓ Should SS_SIP_NO_TIMER_REINVITE_INTERVAL be higher or lower than SS_SIP_SESSION_TTL?

💡 It should be significantly higher. The default SS_SIP_SESSION_TTL is 600 seconds (10 minutes) — this is short because VOS3000 actively probes the call and can detect dead sessions quickly. The default SS_SIP_NO_TIMER_REINVITE_INTERVAL is 7200 seconds (2 hours) — this is much longer because VOS3000 cannot actively verify non-timer calls, so a longer limit avoids cutting legitimate long calls. A good rule of thumb is to set the no timer interval to at least 3–6 times the session TTL value.

📞 Need Expert Help with VOS3000 SIP No Timer Call Duration?

🔧 Configuring the VOS3000 SIP no timer call duration correctly is essential for preventing revenue loss from runaway calls and ensuring billing accuracy. Misconfiguration can lead to either premature call termination or expensive zombie calls.

💬 WhatsApp: +8801911119966 — Get instant expert support for VOS3000 SIP no timer call duration configuration, session timer setup, and complete VoIP network optimization.

📚 Related Resources

• 🔗 VOS3000 Session Timer — Active session timer configuration (RFC 4028)
• 🔗 VOS3000 Max Call Duration — System-level hard call duration limits
• 🔗 VOS3000 SIP Call Flow — Complete SIP signaling lifecycle
• 🔗 VOS3000 Parameter Description — Full parameter reference guide
• 🔗 VOS3000 System Parameters — System-level configuration including SS_MAX_CALL_DURATION
• 🔗 VOS3000 Billing System — Complete billing configuration guide
• 🔗 VOS3000 CDR Billing Discrepancy — Fix billing accuracy issues
• 🔗 VOS3000 SIP Session — SIP session management overview
• 🔗 VOS3000 Call Routing — Call routing and termination configuration
• 🔗 VOS3000 Call Termination — Call termination and hangup reasons
• 🔗 VOS3000 Troubleshooting Guide — Diagnose and fix common issues
• 🔗 VOS3000 Installation — Fresh installation guide
• 🔗 VOS3000 Official Downloads — Official software and documentation

📞 Still have questions about the VOS3000 SIP no timer call duration? Reach out on WhatsApp at +8801911119966 — we provide professional VOS3000 installation, configuration, and support services worldwide. 🌐

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