VOS3000 SIP Authentication Retry, VOS3000 SIP Early Hangup, VOS3000 SIP Session Timer Refresh, VOS3000 Non-Timer Endpoint Safety, VOS3000 SIP NAT Keepalive, VOS3000 SIP Resend Interval, VOS3000 SIP INVITE Timeout, VOS3000 SIP Call Progress Timeout, VOS3000 SIP Outbound Registration Parameters, VOS3000 SIP Privacy Header, VOS3000 SIP Routing Gateway Contact, VOS3000 SIP Publish Expire, VOS3000 SIP Display From, VOS3000 SIP Send Unregister

VOS3000 SIP No Timer Call Duration: Important Maximum Limit Easy Guide

VOS3000 SIP No Timer Call Duration: Important Maximum Limit Guide

๐Ÿ“ž Have you ever discovered runaway calls in your CDR records โ€” 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. โฑ๏ธ

๐Ÿšจ 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 โ€” 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. ๐Ÿ”

๐ŸŽฏ This guide covers everything about the VOS3000 SIP no timer call duration โ€” 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.

Table of Contents

๐Ÿ” What Is VOS3000 SIP No Timer Call Duration?

โฐ 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.

๐Ÿ’ก 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:

  • โŒ No re-INVITE or UPDATE messages can be sent to verify the session
  • โŒ VOS3000 cannot detect whether the far end is still alive
  • โš ๏ธ The only protection is a hard timeout โ€” once exceeded, the call is forcibly terminated
  • ๐Ÿ›ก๏ธ Without this parameter, zombie calls could persist indefinitely

๐Ÿ“ Location in VOS3000 Client: Navigation โ†’ Operation management โ†’ Softswitch management โ†’ Additional settings โ†’ SIP parameter

๐Ÿ“‹ Official Parameter Specification

๐Ÿ”ง According to the VOS3000 2.1.9.07 Official Manual (Table 4-3, Section 4.3.5.2):

AttributeValue
๐Ÿ“Œ Parameter NameSS_SIP_NO_TIMER_REINVITE_INTERVAL
๐Ÿ”ข Default Value7200
๐Ÿ“ UnitSeconds
๐Ÿ“ DescriptionMaximum Conversation Time for Non-TIMER SIP Caller. If SIP caller doesn’t support “timer”, softswitch will stop the call when the time is up.

โฑ๏ธ 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 โ€” regardless of whether the call is still active or has become a zombie.

๐Ÿ”„ VOS3000 SIP No Timer Call Duration vs. Session Timer

๐Ÿ“Š 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:

AspectSession Timer (RFC 4028)No Timer Call Duration
๐Ÿ“Œ ParameterSS_SIP_SESSION_TTLSS_SIP_NO_TIMER_REINVITE_INTERVAL
๐Ÿ”ข Default600s (10 min)7200s (2 hours)
๐ŸŽฏ Applies WhenCaller supports “timer”Caller does NOT support “timer”
๐Ÿ“ก Detection MethodActive โ€” sends re-INVITE/UPDATEPassive โ€” hard timeout only
๐Ÿ” Session-Expires HeaderPresent in SIP messagesNot present
๐Ÿ“ž VerificationPeriodic refresh with 200 OKNone โ€” just countdown
โŒ Call TerminationNo 200 OK โ†’ BYE sentTime exceeded โ†’ BYE sent
๐Ÿ›ก๏ธ Protection LevelHigh โ€” active probingLower โ€” passive timeout

๐Ÿ’ก 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.

๐ŸŽฏ How VOS3000 Decides Which Mechanism to Use

๐Ÿ–ฅ๏ธ When a SIP INVITE arrives at VOS3000, the softswitch inspects the SIP headers to determine whether the caller supports session timers:

๐Ÿ“ž SIP INVITE Arrives at VOS3000
    โ”‚
    โ”œโ”€โ”€ VOS3000 checks for Session-Expires header
    โ”‚
    โ”œโ”€โ”€ โœ… Session-Expires header FOUND
    โ”‚   โ”œโ”€โ”€ Caller supports RFC 4028 session timer
    โ”‚   โ”œโ”€โ”€ VOS3000 uses SS_SIP_SESSION_TTL (default: 600s)
    โ”‚   โ”œโ”€โ”€ Active probing with re-INVITE/UPDATE messages
    โ”‚   โ””โ”€โ”€ Call verified every TTL/Segment interval
    โ”‚
    โ””โ”€โ”€ โŒ Session-Expires header NOT FOUND
        โ”œโ”€โ”€ Caller does NOT support session timer
        โ”œโ”€โ”€ VOS3000 uses SS_SIP_NO_TIMER_REINVITE_INTERVAL (default: 7200s)
        โ”œโ”€โ”€ NO active probing โ€” passive countdown only
        โ””โ”€โ”€ Call forcibly terminated when time exceeds limit

โš™๏ธ SS_SIP_NO_TIMER_REINVITE_INTERVAL โ€” Deep Dive

๐Ÿ” Let’s examine the VOS3000 SIP no timer call duration parameter in full detail โ€” what it does, how it works, and what happens when the limit is reached.

๐Ÿ”‘ How the Parameter Works

โฑ๏ธ When a SIP caller that does not support session timers establishes a call through VOS3000:

  1. ๐Ÿ“ž The call is established normally (INVITE โ†’ 200 OK โ†’ ACK)
  2. ๐Ÿ–ฅ๏ธ VOS3000 detects the absence of a Session-Expires header
  3. โฐ VOS3000 starts a countdown timer set to SS_SIP_NO_TIMER_REINVITE_INTERVAL seconds
  4. ๐Ÿ“Š The call proceeds normally while the countdown runs
  5. ๐Ÿšจ When the countdown reaches zero, VOS3000 sends a BYE message to terminate the call

โš ๏ธ 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 โ€” it cannot detect whether the call is still alive before the timeout.

๐Ÿ“Š Duration Conversion Table

๐Ÿ“‹ Common SS_SIP_NO_TIMER_REINVITE_INTERVAL values and their equivalent durations:

SecondsMinutesHoursCommon Name
900150.25Quarter hour
1800300.5Half hour
3600601One hour
5400901.5Ninety minutes
72001202โœ… Default (two hours)
108001803Three hours
144002404Four hours

๐Ÿ›ก๏ธ Preventing Runaway Calls with VOS3000 SIP No Timer Call Duration

๐Ÿšจ 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 โ€” typically because of network failures, endpoint crashes, or NAT timeouts that prevent proper BYE messages.

โš ๏ธ How Runaway Calls Happen

๐Ÿ“ž Here’s the scenario that creates runaway calls on non-timer endpoints:

๐Ÿ“ž Call Established Between Non-Timer Endpoint and VOS3000
    โ”‚
    โ”œโ”€โ”€ Both parties talk normally
    โ”‚
    โ”œโ”€โ”€ ๐Ÿ”ด Network failure / endpoint crash / NAT timeout
    โ”‚   โ”œโ”€โ”€ No BYE message sent (endpoint is dead/unreachable)
    โ”‚   โ”œโ”€โ”€ Call remains in "connected" state on VOS3000
    โ”‚   โ””โ”€โ”€ VOS3000 CANNOT send re-INVITE (endpoint has no timer support)
    โ”‚
    โ”œโ”€โ”€ โฐ Without SS_SIP_NO_TIMER_REINVITE_INTERVAL:
    โ”‚   โ””โ”€โ”€ โŒ Call stays connected INDEFINITELY
    โ”‚       โ””โ”€โ”€ ๐Ÿ’ธ Billing continues to accumulate
    โ”‚
    โ””โ”€โ”€ โœ… With SS_SIP_NO_TIMER_REINVITE_INTERVAL = 7200s:
        โ””โ”€โ”€ After 2 hours, VOS3000 sends BYE
            โ””โ”€โ”€ ๐Ÿ›ก๏ธ Call terminated, billing stops

๐Ÿ’ก 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.

๐Ÿ“Š Runaway Call Cost Impact Table

๐Ÿ’ธ Understanding the financial impact of runaway calls shows why the VOS3000 SIP no timer call duration setting matters:

Zombie Call DurationRate ($/min)Cost per Incident10 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

๐Ÿšจ 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.

๐Ÿ“Š VOS3000 SIP No Timer Call Duration and Billing Accuracy

๐Ÿ’ฐ Billing accuracy is directly affected by the VOS3000 SIP no timer call duration setting. Here’s how:

๐Ÿ” Billing Impact Analysis

NO_TIMER_INTERVALMax Zombie DurationBilling RiskCDR Accuracy
900s (15 min)15 minutes max๐Ÿ›ก๏ธ Very Lowโœ… Excellent
1800s (30 min)30 minutes maxโœ… Lowโœ… Very Good
3600s (1 hour)1 hour max๐Ÿ”ง Medium-Low๐Ÿ“Š Good
7200s (2 hours) โœ…2 hours maxโš ๏ธ Medium๐Ÿ“Š Acceptable
14400s (4 hours)4 hours max๐Ÿšจ HighโŒ Poor
Not configuredUnlimited๐Ÿ”ฅ CriticalโŒ Very Poor

๐Ÿ“ 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.

๐Ÿ”ง Step-by-Step Configuration of VOS3000 SIP No Timer Call Duration

๐Ÿ–ฅ๏ธ Follow these steps to configure SS_SIP_NO_TIMER_REINVITE_INTERVAL in your VOS3000 softswitch:

Step 1: Navigate to SIP Parameters ๐Ÿ“‹

  1. ๐Ÿ” Log in to VOS3000 Client with administrator credentials
  2. ๐Ÿ“Œ Navigate: Operation management โ†’ Softswitch management โ†’ Additional settings โ†’ SIP parameter
  3. ๐Ÿ” Locate SS_SIP_NO_TIMER_REINVITE_INTERVAL in the SIP parameter list

Step 2: Choose Your Value โฑ๏ธ

๐ŸŽฏ Select the appropriate value based on your deployment type:

Deployment TypeRecommended ValueDurationRationale
๐Ÿข Standard enterprise7200s2 hoursโœ… Default โ€” sufficient for most calls
๐Ÿ“ž Wholesale termination3600s1 hour๐Ÿ”ง Tighter control, lower risk
๐Ÿ›ก๏ธ Premium / high-value routes1800s30 minutes๐Ÿ” Maximum billing protection
๐ŸŒ Legacy gateway networks1800sโ€“3600s30โ€“60 min๐Ÿ“ก Old devices often lack timer support
๐Ÿ“ž Call center operations5400s90 minutes๐Ÿ“Š Accommodates long agent calls
๐Ÿ”ฅ Maximum protection900s15 minutes๐Ÿ›ก๏ธ Zero tolerance for runaway calls

Step 3: Apply and Save โœ…

  1. ๐Ÿ“ Enter the desired value (in seconds) in the SS_SIP_NO_TIMER_REINVITE_INTERVAL field
  2. ๐Ÿ’พ Click Save to apply the configuration
  3. ๐Ÿ”„ The new value takes effect for all subsequent calls from non-timer SIP endpoints

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

๐Ÿ”„ Relationship with Other VOS3000 Parameters

๐Ÿ”— 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:

ParameterDefaultUnitRelationship to NO_TIMER
SS_SIP_SESSION_TTL600Seconds๐Ÿ”„ Complementary โ€” applies when timer IS supported
SS_SIP_SESSION_UPDATE_SEGMENT2Count๐Ÿ“Š Controls re-INVITE frequency for timer calls
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP0Secondsโฐ Grace period โ€” applies only to timer calls
SS_MAX_CALL_DURATIONNoneโ€”๐Ÿ›ก๏ธ System-level hard limit for ALL calls

๐Ÿ’ก 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.

๐Ÿ“‹ Parameter Interaction Flow

๐Ÿ“ž Call Arrives at VOS3000
    โ”‚
    โ”œโ”€โ”€ Check: Does SS_MAX_CALL_DURATION exist?
    โ”‚   โ”œโ”€โ”€ YES โ†’ Apply system-level hard limit
    โ”‚   โ””โ”€โ”€ NO  โ†’ No system-level limit
    โ”‚
    โ”œโ”€โ”€ Check: Does caller support "timer"?
    โ”‚   โ”œโ”€โ”€ YES โ†’ Apply SS_SIP_SESSION_TTL (600s default)
    โ”‚   โ”‚        Active probing via re-INVITE/UPDATE
    โ”‚   โ”‚        Hang up if no 200 OK confirmation
    โ”‚   โ”‚
    โ”‚   โ””โ”€โ”€ NO  โ†’ Apply SS_SIP_NO_TIMER_REINVITE_INTERVAL (7200s default)
    โ”‚            NO active probing โ€” passive countdown
    โ”‚            Hang up when time exceeded
    โ”‚
    โ””โ”€โ”€ ๐Ÿ›ก๏ธ Effective limit = min(SS_MAX_CALL_DURATION, applicable timer)

๐Ÿ’ก Best Practices for VOS3000 SIP No Timer Call Duration

๐ŸŽฏ Follow these best practices to maximize the effectiveness of your VOS3000 SIP no timer call duration configuration:

Best PracticeRecommendationReason
๐ŸŽฏ Set SS_MAX_CALL_DURATIONConfigure a system-level limit as backup๐Ÿ›ก๏ธ Double protection for all calls
๐Ÿ“Š Monitor CDR recordsCheck for calls near the 7200s limit weekly๐Ÿ” Detects non-timer endpoint patterns
๐Ÿ“ž Encourage timer supportAsk vendors to enable RFC 4028 on endpointsโœ… Active probing is far superior
๐Ÿ”ง Lower for premium routesSet 1800sโ€“3600s for expensive destinations๐Ÿ” Minimizes billing exposure
๐Ÿ”„ Coordinate with session timerNO_TIMER should be โ‰ฅ 3ร— SS_SIP_SESSION_TTL๐Ÿ“Š Consistent protection across both modes
๐Ÿ“ Document configurationRecord all timer-related parameter values๐Ÿ“‹ Simplifies troubleshooting later
๐Ÿ“ก Verify endpoint compatibilityCapture SIP INVITE to check Session-Expires๐Ÿ” Confirms which mode is active

๐Ÿ’ก 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 1800sโ€“3600s for better protection. Check our VOS3000 parameter description guide for the complete parameter reference.

๐Ÿ›ก๏ธ Common Problems and Troubleshooting

โš ๏ธ Here are the most common issues related to the VOS3000 SIP no timer call duration and their solutions:

โŒ Problem 1: Calls Being Cut After Exactly 2 Hours

๐Ÿ” Symptom: Legitimate long-duration calls are being terminated at exactly 2 hours.

๐Ÿ’ก Cause: The SIP caller does not support session timers, and SS_SIP_NO_TIMER_REINVITE_INTERVAL is set to the default 7200 seconds.

โœ… Solutions:

  • ๐Ÿ”ง Increase SS_SIP_NO_TIMER_REINVITE_INTERVAL if 2-hour calls are expected
  • ๐Ÿ“ž Ask the SIP endpoint vendor to implement RFC 4028 session timer support
  • ๐Ÿ” Verify the call flow using our SIP call flow guide

โŒ Problem 2: Ultra-Long Bills from Non-Timer Endpoints

๐Ÿ” Symptom: CDR records show calls lasting the full 7200 seconds, but the actual conversation was much shorter.

๐Ÿ’ก Cause: The endpoint crashed or lost network connectivity without sending BYE, and the non-timer interval is too long.

โœ… Solutions:

  • โฑ๏ธ Reduce SS_SIP_NO_TIMER_REINVITE_INTERVAL to 1800s or 3600s
  • ๐Ÿ›ก๏ธ Set SS_MAX_CALL_DURATION as a secondary safety limit
  • ๐Ÿ“Š Cross-reference CDR records with billing system data

โŒ Problem 3: Not Sure Which Endpoints Support Session Timers

๐Ÿ” Symptom: Unknown whether your SIP trunks and gateways support RFC 4028.

๐Ÿ’ก Solution: Capture the SIP INVITE message and check for the Session-Expires header:

# SIP INVITE from a TIMER-capable endpoint:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060
Session-Expires: 600           <-- โœ… Timer SUPPORTED
Min-SE: 90
...

# SIP INVITE from a NON-TIMER endpoint:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060
                                <-- โŒ No Session-Expires header
...
# VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL for this call

๐Ÿ“ž Need more help with SIP debugging? See our VOS3000 troubleshooting guide for detailed instructions.

๐Ÿ“Š Complete VOS3000 SIP No Timer Call Duration Decision Matrix

๐ŸŽฏ Use this decision matrix to select the optimal SS_SIP_NO_TIMER_REINVITE_INTERVAL value for your deployment:

FactorLow Value (900โ€“1800s)Mid Value (3600โ€“5400s)High Value (7200s+)
๐Ÿ›ก๏ธ Billing riskโœ… Very low๐Ÿ”ง Moderateโš ๏ธ Higher
๐Ÿ“ž Call disruptionโš ๏ธ Possible for long callsโœ… Rareโœ… Very rare
๐Ÿ’ธ Zombie call costโœ… Minimal๐Ÿ”ง Controlledโš ๏ธ Potentially high
๐Ÿ“Š CDR accuracyโœ… Excellent๐Ÿ“Š Good๐Ÿ”ง Acceptable
๐ŸŽฏ Best forPremium routes, high ratesWholesale, mixed trafficStandard enterprise, low rates

โ“ Frequently Asked Questions

โ“ What is the default VOS3000 SIP no timer call duration?

โฑ๏ธ 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).

โ“ What happens when VOS3000 SIP no timer call duration is exceeded?

๐Ÿšจ 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 โ€” there is no grace period or retry mechanism for non-timer calls.

โ“ How is VOS3000 SIP no timer call duration different from session timer?

๐Ÿ”„ 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 โ€” 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.

โ“ Can I set SS_SIP_NO_TIMER_REINVITE_INTERVAL to unlimited?

โŒ 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.

โ“ Does VOS3000 SIP no timer call duration affect calls that support session timers?

๐Ÿ“ฑ 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 โ€” each call uses one or the other based on the endpoint’s timer support.

โ“ How do I check if my SIP endpoints support session timers?

๐Ÿ” 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.

โ“ Should SS_SIP_NO_TIMER_REINVITE_INTERVAL be higher or lower than SS_SIP_SESSION_TTL?

๐Ÿ’ก It should be significantly higher. The default SS_SIP_SESSION_TTL is 600 seconds (10 minutes) โ€” 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) โ€” 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 3โ€“6 times the session TTL value.

๐Ÿ“ž Need Expert Help with VOS3000 SIP No Timer Call Duration?

๐Ÿ”ง 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.

๐Ÿ’ฌ WhatsApp: +8801911119966 โ€” Get instant expert support for VOS3000 SIP no timer call duration configuration, session timer setup, and complete VoIP network optimization.

๐Ÿ“ž Still have questions about the VOS3000 SIP no timer call duration? Reach out on WhatsApp at +8801911119966 โ€” we provide professional VOS3000 installation, configuration, and support services worldwide. ๐ŸŒ


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

๐Ÿ“ฑ WhatsApp: +8801911119966
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VOS3000 SIP Authentication Retry, VOS3000 SIP Early Hangup, VOS3000 SIP Session Timer Refresh, VOS3000 Non-Timer Endpoint Safety, VOS3000 SIP NAT KeepaliveVOS3000 SIP Authentication Retry, VOS3000 SIP Early Hangup, VOS3000 SIP Session Timer Refresh, VOS3000 Non-Timer Endpoint Safety, VOS3000 SIP NAT KeepaliveVOS3000 SIP Authentication Retry, VOS3000 SIP Early Hangup, VOS3000 SIP Session Timer Refresh, VOS3000 Non-Timer Endpoint Safety, VOS3000 SIP NAT Keepalive
VOS3000 SIP Authentication Retry, VOS3000 SIP Early Hangup, VOS3000 SIP Session Timer Refresh, VOS3000 Non-Timer Endpoint Safety, VOS3000 SIP NAT Keepalive, VOS3000 SIP Resend Interval, VOS3000 SIP INVITE Timeout, VOS3000 SIP Call Progress Timeout, VOS3000 SIP Outbound Registration Parameters, VOS3000 SIP Privacy Header, VOS3000 SIP Routing Gateway Contact, VOS3000 SIP Publish Expire, VOS3000 SIP Display From, VOS3000 SIP Send Unregister

VOS3000 SIP Session Timer: Powerful RFC 4028 Setup Guide

VOS3000 SIP Session Timer: Powerful RFC 4028 Setup Guide

๐Ÿ“ž Are mysterious ghost calls and ultra-long bills draining your VoIP revenue? The VOS3000 SIP session timer is your first line of defense. Based on RFC 4028, this critical SIP protocol feature detects whether calls are still alive โ€” and automatically hangs up dead sessions before they inflate your billing. โฑ๏ธ

๐Ÿ”ง In abnormal network conditions, SIP endpoints can lose connectivity without sending a proper BYE message. Without session timers, these zombie calls linger indefinitely, generating charges for conversations that ended long ago. VOS3000 solves this with four powerful parameters that control how session timers operate across your entire softswitch.

๐ŸŽฏ This guide walks you through every VOS3000 SIP session timer parameter โ€” from SS_SIP_SESSION_TTL to SS_SIP_NO_TIMER_REINVITE_INTERVAL โ€” with real default values, configuration steps, and best practices to keep your VoIP network clean and profitable.

Table of Contents

๐Ÿ” What Is VOS3000 SIP Session Timer?

โฐ The VOS3000 SIP session timer is a built-in mechanism that periodically verifies whether a SIP call is still active. It follows the RFC 4028 SIP Session Timers standard, which defines how SIP User Agents can request, negotiate, and maintain session timers during a call.

๐Ÿ’ก Why it matters: In VoIP networks, network failures, NAT timeouts, and endpoint crashes can leave calls in a “connected” state even after both parties have stopped communicating. The VOS3000 SIP session timer prevents these orphaned calls by:

  • ๐Ÿ”„ Periodically sending re-INVITE or UPDATE messages to confirm the call is still alive
  • โŒ Automatically hanging up calls when no confirmation is received
  • ๐Ÿ›ก๏ธ Preventing ultra-long bills caused by zombie sessions
  • ๐Ÿ“Š Detecting abnormal network conditions in real time

๐Ÿ“ Location in VOS3000 Client: Navigation โ†’ Operation management โ†’ Softswitch management โ†’ Additional settings โ†’ SIP parameter

๐Ÿ“‹ RFC 4028 Core Concepts for VOS3000

๐ŸŒ RFC 4028 introduces the Session-Expires header and Min-SE header to SIP. Here’s how they map to VOS3000:

RFC 4028 ConceptVOS3000 ParameterFunction
Session-ExpiresSS_SIP_SESSION_TTLTotal session lifetime before refresh required
Refresher negotiationSS_SIP_SESSION_UPDATE_SEGMENTNumber of refresh attempts within TTL
Early terminationSS_SIP_SESSION_TIMEOUT_EARLY_HANGUPGrace period before early hangup on no response
Non-timer fallbackSS_SIP_NO_TIMER_REINVITE_INTERVALMax call duration for non-session-timer UAs

โš™๏ธ VOS3000 SIP Session Timer Parameters Deep Dive

๐Ÿ”ง Let’s examine each parameter in detail using the official VOS3000 2.1.9.07 manual data.

๐Ÿ”‘ SS_SIP_SESSION_TTL โ€” Detecting SIP Connected Status Interval

โฑ๏ธ SS_SIP_SESSION_TTL is the heart of the VOS3000 SIP session timer system. It defines the total interval (in seconds) within which VOS3000 will detect whether a SIP call is still connected.

AttributeValue
๐Ÿ“Œ Parameter NameSS_SIP_SESSION_TTL
๐Ÿ”ข Default Value600 seconds (10 minutes)
๐Ÿ“ UnitSeconds
๐Ÿ“ DescriptionIf SIP caller supports “session-timer”, within the time softswitch will detect connect status according to the retry times. If got no confirm message, softswitch will regard as call finish, then hang up.

๐Ÿ’ก How it works: When a SIP caller that supports session-timer establishes a call, VOS3000 starts a countdown based on SS_SIP_SESSION_TTL. Within this period, VOS3000 divides the TTL into segments (controlled by SS_SIP_SESSION_UPDATE_SEGMENT) and sends re-INVITE or UPDATE messages at each segment boundary. If no confirmation comes back, the call is terminated.

โš ๏ธ Setting too low: A TTL of 60 seconds means frequent re-INVITEs, increasing signaling overhead. Setting too high: A TTL of 3600 seconds means zombie calls can persist for up to an hour. The default of 600 seconds (10 minutes) strikes a practical balance.

๐Ÿ”„ SS_SIP_SESSION_UPDATE_SEGMENT โ€” Reinvite Interval Divider

๐Ÿ“Š SS_SIP_SESSION_UPDATE_SEGMENT controls how many times VOS3000 will attempt to refresh a session within the TTL period. It directly determines the re-INVITE or UPDATE interval.

AttributeValue
๐Ÿ“Œ Parameter NameSS_SIP_SESSION_UPDATE_SEGMENT
๐Ÿ”ข Default Value2
๐Ÿ“ Range2 โ€“ 10
๐Ÿ“ DescriptionSIP Timer reinvite (update) Interval โ€” divides the TTL into segments

๐ŸŽฏ Calculation: The actual re-INVITE interval = SS_SIP_SESSION_TTL รท SS_SIP_SESSION_UPDATE_SEGMENT

TTL (seconds)SegmentRe-INVITE IntervalUse Case
6002300s (5 min)โœ… Default โ€” balanced
6004150s (2.5 min)๐Ÿ”ง More frequent checks
6006100s (1.7 min)๐Ÿ“ก Unstable networks
6001060s (1 min)โš ๏ธ High overhead
18003600s (10 min)๐Ÿ“ž Long calls, stable net

๐Ÿ’ก Key insight: With the default settings (TTL=600, Segment=2), VOS3000 sends a re-INVITE every 300 seconds (5 minutes). If the far end responds with 200 OK, the session is confirmed alive. If not, the call is hung up.

โฐ SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP โ€” Early Hangup Timer

๐Ÿ”’ SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP adds a safety net by specifying how many seconds to wait before performing an early hangup when a re-INVITE or UPDATE receives no response.

AttributeValue
๐Ÿ“Œ Parameter NameSS_SIP_SESSION_TIMEOUT_EARLY_HANGUP
๐Ÿ”ข Default Value0 seconds (disabled)
๐Ÿ“ UnitSeconds
๐Ÿ“ DescriptionSIP Timer no reinvite (update) Early Hang up โ€” extra grace period before terminating

โš ๏ธ When set to 0 (default): VOS3000 hangs up immediately when the session timer expires without confirmation. No grace period is given.

โœ… When set to a positive value: VOS3000 waits the specified number of seconds after the timer expires before hanging up. This gives the far end a brief window to recover from momentary network glitches.

๐Ÿ’ก Recommended setting: For most deployments, keep at 0 for immediate cleanup. On networks with occasional packet loss, set to 5-10 seconds for a small grace window.

๐Ÿ–ฅ๏ธ SS_SIP_NO_TIMER_REINVITE_INTERVAL โ€” Non-Timer SIP Caller Limit

๐Ÿ“ฑ Not all SIP endpoints support session timers. SS_SIP_NO_TIMER_REINVITE_INTERVAL handles this scenario by setting a maximum conversation time for SIP callers that do NOT support the “timer” feature.

AttributeValue
๐Ÿ“Œ Parameter NameSS_SIP_NO_TIMER_REINVITE_INTERVAL
๐Ÿ”ข Default Value7200 seconds (2 hours)
๐Ÿ“ UnitSeconds
๐Ÿ“ DescriptionIf SIP caller doesn’t support “timer”, softswitch will stop the call when the time is up

๐Ÿ” Critical function: Since non-timer SIP callers cannot respond to session refresh requests, VOS3000 cannot actively verify if the call is still alive. The only protection is a hard timeout โ€” once the call duration exceeds this value, VOS3000 forcibly terminates it.

โš ๏ธ Default of 7200s (2 hours): This means a zombie call from a non-timer endpoint could persist for up to 2 hours. For high-value routes, consider lowering this to 3600s (1 hour) or even 1800s (30 minutes).

๐Ÿ“‹ How VOS3000 SIP Session Timer Works โ€” Complete Flow

๐Ÿ”„ Understanding the full session timer flow is essential for proper configuration. Here’s exactly what happens during a call:

๐ŸŽฏ Scenario A: Caller SUPPORTS Session Timer

๐Ÿ“ž Call Established (200 OK)
    โ”‚
    โ”œโ”€โ”€ VOS3000 starts TTL countdown (SS_SIP_SESSION_TTL = 600s)
    โ”‚
    โ”œโ”€โ”€ At TTL/Segment = 300s โ”€โ”€โ–บ VOS3000 sends re-INVITE/UPDATE
    โ”‚   โ”œโ”€โ”€ โœ… 200 OK received โ†’ Session confirmed, timer resets
    โ”‚   โ””โ”€โ”€ โŒ No response โ†’ Retry at next segment
    โ”‚
    โ”œโ”€โ”€ At TTL = 600s โ”€โ”€โ–บ Final check
    โ”‚   โ”œโ”€โ”€ โœ… 200 OK received โ†’ Session confirmed, timer resets
    โ”‚   โ””โ”€โ”€ โŒ No response โ†’ Call terminated (BYE sent)
    โ”‚       โ””โ”€โ”€ If EARLY_HANGUP > 0 โ†’ Wait X seconds, then BYE
    โ”‚
    โ””โ”€โ”€ ๐Ÿ” Cycle repeats for duration of call

๐ŸŽฏ Scenario B: Caller Does NOT Support Session Timer

๐Ÿ“ž Call Established (200 OK โ€” no Session-Expires header)
    โ”‚
    โ”œโ”€โ”€ VOS3000 detects no timer support
    โ”‚
    โ”œโ”€โ”€ No re-INVITE/UPDATE messages sent
    โ”‚
    โ”œโ”€โ”€ Call continues until...
    โ”‚   โ”œโ”€โ”€ ๐Ÿ“ฑ Normal BYE from either party, OR
    โ”‚   โ””โ”€โ”€ โฐ Duration exceeds SS_SIP_NO_TIMER_REINVITE_INTERVAL (7200s)
    โ”‚       โ””โ”€โ”€ VOS3000 forcibly terminates call (BYE sent)
    โ”‚
    โ””โ”€โ”€ โŒ No active session detection possible

๐Ÿ”ง Step-by-Step VOS3000 SIP Session Timer Configuration

๐Ÿ–ฅ๏ธ Follow these steps to configure the VOS3000 SIP session timer parameters:

Step 1: Navigate to SIP Parameters ๐Ÿ“‹

  1. ๐Ÿ” Log in to VOS3000 Client
  2. ๐Ÿ“Œ Navigate: Operation management โ†’ Softswitch management โ†’ Additional settings โ†’ SIP parameter
  3. ๐Ÿ” Locate the session timer parameters in the parameter list

Step 2: Configure SS_SIP_SESSION_TTL โฑ๏ธ

Deployment TypeRecommended TTLRationale
๐Ÿข Standard enterprise600s (default)โœ… Good balance of detection and overhead
๐Ÿ“ž High-volume wholesale300s โ€“ 600s๐Ÿ”ง Faster zombie detection on busy routes
๐ŸŒ Unstable networks180s โ€“ 300s๐Ÿ“ก Quick detection of dropped calls
๐Ÿ›ก๏ธ Premium routes900s โ€“ 1800s๐Ÿ” Less signaling overhead, longer calls OK

Step 3: Set SS_SIP_SESSION_UPDATE_SEGMENT ๐Ÿ”„

๐Ÿ“Š Choose the segment value based on your network reliability:

Segment ValueTTL=600 IntervalRetry CountBest For
2 (default)300s2 attemptsโœ… Most deployments
3200s3 attempts๐Ÿ”ง Moderate reliability
5120s5 attempts๐Ÿ“ก Flaky connections
875s8 attemptsโš ๏ธ Very unstable nets

Step 4: Configure Early Hangup โฐ

๐Ÿ”’ Set SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP based on your tolerance for ghost calls:

  • โœ… 0 seconds (default): Immediate hangup โ€” zero tolerance for zombie calls
  • ๐Ÿ”ง 5-10 seconds: Small grace window for momentary network blips
  • โš ๏ธ 30+ seconds: Not recommended โ€” defeats the purpose of session timers

Step 5: Adjust Non-Timer Caller Limit ๐Ÿ“ฑ

๐ŸŽฏ Set SS_SIP_NO_TIMER_REINVITE_INTERVAL based on your risk tolerance:

SettingDurationRisk LevelUse Case
7200s (default)2 hoursโš ๏ธ MediumStandard VoIP operations
3600s1 hour๐Ÿ”ง Low-MediumWholesale termination
1800s30 minutesโœ… LowHigh-value premium routes
900s15 minutes๐Ÿ›ก๏ธ Very LowMaximum protection

๐Ÿ“Š Complete VOS3000 SIP Session Timer Parameter Reference

๐Ÿ“‹ Here’s the full reference table combining all session timer parameters from the official VOS3000 2.1.9.07 manual:

ParameterDefaultUnitRangePurpose
SS_SIP_SESSION_TTL600Secondsโ€”Session expiry detection interval
SS_SIP_SESSION_UPDATE_SEGMENT2Count2โ€“10Re-INVITE interval divider
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP0Secondsโ€”Grace period before early hangup
SS_SIP_NO_TIMER_REINVITE_INTERVAL7200Secondsโ€”Max call time for non-timer UAs

๐Ÿ›ก๏ธ Common VOS3000 SIP Session Timer Problems and Solutions

โš ๏ธ Even with proper configuration, session timer issues can arise. Here are the most common problems and their fixes:

โŒ Problem 1: Calls Dropping Every 5 Minutes

๐Ÿ” Symptom: Active calls are being terminated at exactly the re-INVITE interval.

๐Ÿ’ก Cause: The far-end SIP device does not properly respond to re-INVITE or UPDATE messages. The VOS3000 SIP session timer interprets the lack of response as a dead call.

โœ… Solutions:

  • ๐Ÿ”ง Increase SS_SIP_SESSION_TTL to give more time per cycle
  • ๐Ÿ”„ Reduce SS_SIP_SESSION_UPDATE_SEGMENT for fewer but longer intervals
  • ๐Ÿ“ก Verify the far-end device supports RFC 4028 session timers
  • ๐Ÿ“ž Check if the far-end is behind a SIP ALG that drops re-INVITEs โ€” see our SIP debug guide

โŒ Problem 2: Ultra-Long Bills from Zombie Calls

๐Ÿ” Symptom: CDR records show calls lasting hours beyond actual conversation time.

๐Ÿ’ก Cause: The SIP caller does not support session timers, and SS_SIP_NO_TIMER_REINVITE_INTERVAL is too high.

โœ… Solutions:

  • โฑ๏ธ Reduce SS_SIP_NO_TIMER_REINVITE_INTERVAL from 7200 to 1800 or lower
  • ๐Ÿ” Ensure SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP is set to 0 (immediate cleanup)
  • ๐Ÿ“Š Monitor CDR records for abnormally long calls โ€” use our CDR billing discrepancy guide

โŒ Problem 3: Excessive Signaling Overhead

๐Ÿ” Symptom: High CPU usage on VOS3000 server, excessive SIP signaling traffic.

๐Ÿ’ก Cause: SS_SIP_SESSION_UPDATE_SEGMENT is set too high, causing frequent re-INVITEs.

โœ… Solutions:

  • ๐Ÿ“Š Reduce SS_SIP_SESSION_UPDATE_SEGMENT to 2 (default) for fewer refresh attempts
  • โฑ๏ธ Increase SS_SIP_SESSION_TTL to 900 or 1800 for longer cycles
  • ๐Ÿ”ง Balance detection speed against signaling load

๐Ÿ’ก VOS3000 SIP Session Timer Best Practices

๐ŸŽฏ Follow these best practices to get the most from your VOS3000 SIP session timer configuration:

Best PracticeRecommendationReason
๐ŸŽฏ Start with defaultsTTL=600, Segment=2Proven balance for most deployments
๐Ÿ“Š Monitor CDRsCheck for abnormally long calls weeklyDetects zombie calls early
๐Ÿ”’ Lower non-timer limitSet NO_TIMER to 1800โ€“3600Reduces risk from non-RFC 4028 endpoints
๐Ÿ”„ Test before productionVerify with SIP debug toolsAvoids unexpected call drops
๐Ÿ“ž Verify endpoint supportCheck Session-Expires in SIP INVITEConfirms timer negotiation works
๐Ÿ›ก๏ธ Keep early hangup at 0Unless network is very unstableImmediate cleanup is safer

๐Ÿ’ก Pro tip: The VOS3000 SIP session timer works hand-in-hand with your max call duration settings. While session timers actively detect dead calls, the max call duration parameter enforces a hard limit on all calls regardless of their state. Configure both for maximum protection.

๐Ÿ”„ VOS3000 SIP Session Timer and SIP Call Flow Interaction

๐Ÿ“ก The session timer operates within the broader SIP call flow. Understanding how it interacts with other SIP messages is critical:

๐Ÿ“ฑ SIP Call Flow with Session Timer:

Caller โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€ VOS3000 โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€ Called Party
  โ”‚                              โ”‚                              โ”‚
  โ”‚โ”€โ”€โ”€โ”€ INVITE โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ–บโ”‚โ”€โ”€โ”€โ”€ INVITE โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ–บโ”‚
  โ”‚   (Session-Expires: 600)    โ”‚   (Session-Expires: 600)    โ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚โ—„โ”€โ”€โ”€ 200 OK โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”‚โ—„โ”€โ”€โ”€ 200 OK โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”‚
  โ”‚   (Session-Expires: 600)    โ”‚   (Session-Expires: 600)    โ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚       ... call in progress ...                              โ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚      โ”Œโ”€ TTL/Segment timer โ”€โ”€โ”                              โ”‚
  โ”‚      โ”‚  (300s elapsed)      โ”‚                              โ”‚
  โ”‚      โ””โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”˜                              โ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚โ—„โ”€โ”€โ”€ re-INVITE/UPDATE โ”€โ”€โ”€โ”€โ”€โ”€โ”‚โ”€โ”€โ”€โ”€ re-INVITE/UPDATE โ”€โ”€โ”€โ”€โ”€โ”€โ–บโ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚โ”€โ”€โ”€โ”€ 200 OK โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ–บโ”‚โ—„โ”€โ”€โ”€ 200 OK โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”‚
  โ”‚                              โ”‚                              โ”‚
  โ”‚       ... timer resets ...                                  โ”‚
  โ”‚                              โ”‚                              โ”‚
  โŒ If no 200 OK response:                                     โ”‚
  โ”‚                              โ”‚โ”€โ”€โ”€โ”€ BYE โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ–บโ”‚
  โ”‚โ—„โ”€โ”€โ”€ BYE โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”€โ”‚                              โ”‚

๐Ÿ”ง For a deeper understanding of how session timers fit into the complete SIP call lifecycle, see our comprehensive SIP call flow guide.

๐Ÿ” Verifying VOS3000 SIP Session Timer Operation

๐Ÿ“ After configuration, verify that session timers are working correctly:

Using SIP Debug to Confirm Timer Negotiation ๐Ÿ”

# Check SIP INVITE for Session-Expires header
# This confirms the caller supports session timers

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060
From: <sip:[email protected]>;tag=abc123
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Session-Expires: 600        <-- ๐Ÿ”‘ Session timer negotiated!
Min-SE: 90                  <-- ๐Ÿ”‘ Minimum session interval
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: ...

# If no Session-Expires header appears,
# the caller does NOT support session timers
# VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL instead

๐Ÿ“ž Need help debugging SIP signaling? Check our SIP debug guide for step-by-step Wireshark capture instructions.

โ“ Frequently Asked Questions

โ“ What is the default VOS3000 SIP session timer value?

โฑ๏ธ The default VOS3000 SIP session timer value is 600 seconds (10 minutes), configured via the SS_SIP_SESSION_TTL parameter. This means VOS3000 will attempt to verify call connectivity every 600 seconds divided by the SS_SIP_SESSION_UPDATE_SEGMENT value (default 2), resulting in a re-INVITE every 300 seconds.

โ“ How does VOS3000 handle SIP callers that do not support session timers?

๐Ÿ“ฑ When a SIP caller does not support the “timer” feature (no Session-Expires header in INVITE/200 OK), VOS3000 cannot send re-INVITE or UPDATE messages to verify the call. Instead, it uses the SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter (default: 7200 seconds / 2 hours) as a hard limit. When the call duration exceeds this value, VOS3000 forcibly terminates the call.

โ“ Can I set SS_SIP_SESSION_UPDATE_SEGMENT to 1?

โŒ No. The valid range for SS_SIP_SESSION_UPDATE_SEGMENT is 2 to 10. A value of 1 would mean only one attempt to verify the session, which provides no retry capability. The minimum of 2 ensures at least one re-INVITE and one retry opportunity within the TTL period.

โ“ What happens when VOS3000 SIP session timer detects a dead call?

๐Ÿ”’ When VOS3000 sends a re-INVITE or UPDATE and receives no 200 OK confirmation within the TTL period, it considers the call finished. VOS3000 then sends a BYE message to terminate the call. If SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP is set to a value greater than 0, VOS3000 will wait that many seconds before sending the BYE, giving the endpoint a brief grace period to recover.

โ“ Is the VOS3000 SIP session timer compliant with RFC 4028?

โœ… Yes. The VOS3000 SIP session timer implementation follows RFC 4028 โ€” Session Timers in the Session Initiation Protocol. VOS3000 supports the Session-Expires header, re-INVITE and UPDATE refresh methods, and proper session timer negotiation as defined in the RFC. Refer to the official VOS3000 documentation at vos3000.com for detailed compliance information.

โ“ Should I enable SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP?

๐Ÿ’ก It depends on your network conditions. The default value of 0 (disabled) is recommended for most deployments because it provides immediate cleanup of dead sessions. If your network experiences occasional momentary packet loss that could cause a re-INVITE response to be delayed by a few seconds, you can set it to 5-10 seconds for a small grace window. Values above 30 seconds are not recommended as they undermine the purpose of session timers.

โ“ How does VOS3000 SIP session timer prevent ultra-long bills?

๐Ÿ›ก๏ธ Ultra-long bills occur when calls remain in “connected” state after the actual conversation has ended โ€” typically due to network failures, NAT timeouts, or endpoint crashes that prevent proper BYE messages. The VOS3000 SIP session timer prevents this by actively probing the call at regular intervals. If the far-end cannot confirm the session is still alive, VOS3000 terminates it. For non-timer endpoints, the SS_SIP_NO_TIMER_REINVITE_INTERVAL enforces a hard maximum duration. Combined with proper billing system configuration, this effectively eliminates zombie-call billing.

๐Ÿ“ž Need Expert Help with VOS3000 SIP Session Timer?

๐Ÿ”ง Configuring the VOS3000 SIP session timer correctly is critical for preventing revenue loss from zombie calls and ultra-long bills. If you need expert assistance with your VOS3000 deployment, our team is ready to help.

๐Ÿ’ฌ WhatsApp: +8801911119966 โ€” Get instant support for VOS3000 SIP session timer configuration, RFC 4028 compliance, and VoIP network optimization.

๐Ÿ“ž Still have questions about the VOS3000 SIP session timer? Reach out on WhatsApp at +8801911119966 โ€” we provide professional VOS3000 installation, configuration, and support services worldwide. ๐ŸŒ


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