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

๐Ÿ“ฑ WhatsApp: +8801911119966
๐ŸŒ Website: www.vos3000.com
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๐Ÿ“ฅ Downloads: VOS3000 Downloads


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 Authentication Suspend, VOS3000 Registration Flood Protection, VOS3000 No Media Hangup, VOS3000 Max Call Duration Limit, VOS3000 Billing Precision

VOS3000 No Media Hangup: Smart Auto-Disconnect for Ghost Calls Important

VOS3000 No Media Hangup: Smart Auto-Disconnect for Ghost Calls

In wholesale VoIP operations, few problems are as insidious and costly as ghost calls โ€” calls that remain connected in SIP signaling but have no RTP media flowing. These phantom sessions silently consume concurrent call capacity, inflate CDR durations, and generate billing disputes that erode customer trust. The VOS3000 no media hangup feature, configured through the SS_NOMEDIAHANGUPTIME system parameter documented in VOS3000 Manual Section 4.3.5.2, provides a Smart automatic disconnect mechanism that monitors RTP streams and terminates calls when media stops flowing for a configurable period.

This comprehensive guide explains what ghost calls are, how they impact your VoIP business, and how to configure VOS3000 no media hangup to automatically clean up dead call sessions. Whether you are dealing with NAT timeout issues, endpoint crashes, or one-way audio scenarios that leave zombie calls on your server, this guide covers the complete configuration, testing, and troubleshooting process. For professional assistance with VOS3000 ghost call prevention, contact us on WhatsApp at +8801911119966.

What Are Ghost Calls in VoIP?

A ghost call is a VoIP session that remains established in SIP signaling but has no active RTP media stream. The SIP dialog is still valid โ€” the call appears as “answered” and “connected” in the system โ€” but no voice packets are flowing between the endpoints. From the VOS3000 softswitch perspective, the call slot is occupied, the CDR timer is running, and the session counts against your concurrent call limit, but there is no actual voice communication happening.

Ghost calls are particularly dangerous because they are invisible to the caller and callee. Neither party is aware that a call session is still open on the server. The SIP signaling path may have been maintained through keepalive messages or simply because neither side sent a BYE message, while the RTP media path has completely died. The result is a zombie call that wastes resources and corrupts billing data until someone or something terminates it.

Why Ghost Calls Are a Serious Problem

Ghost calls create multiple layers of problems for VoIP operators:

  • Wasted concurrent call capacity: Every ghost call occupies a license slot that could be used for a real call. During network instability events, hundreds of ghost calls can accumulate, exhausting your concurrent call capacity and blocking legitimate traffic
  • Incorrect billing: CDR records show the full duration from answer to disconnect, including the period when no media was flowing. Customers are billed for dead air time, leading to disputes and chargebacks
  • Inflated CDR durations: Ghost calls can last for hours because neither endpoint sends a BYE. CDR records show extremely long call durations with no corresponding voice activity, distorting traffic analytics
  • Billing disputes: When customers analyze their CDRs and find calls lasting hours with no conversation, they dispute the charges. Resolving these disputes consumes time and damages business relationships
  • Resource exhaustion: Each ghost call maintains state in the VOS3000 media relay, consuming memory and processing resources that should be available for active calls

For a deeper understanding of VOS3000 media handling, see our VOS3000 RTP media guide.

How Ghost Calls Occur: Causes and Symptoms

Understanding the root causes of ghost calls is essential for effective prevention. Ghost calls typically occur when the SIP signaling path survives while the RTP media path fails. This section covers the most common causes and their telltale symptoms.

๐Ÿ‘ป Cause๐Ÿ“‹ Description๐Ÿ” Symptom in CDRโš ๏ธ Impact Level
Network connectivity lossInternet link failure between VOS3000 and one endpoint; SIP path via alternate route but RTP direct path brokenCall duration extends far beyond normal; no media packets during outage windowHigh โ€” multiple simultaneous ghost calls during outage
NAT timeoutNAT device drops RTP pinhole mapping due to inactivity; SIP signaling on separate pinhole survivesOne-way audio progressing to no audio; call remains connected indefinitelyMedium โ€” affects specific endpoint pairs behind NAT
Endpoint crash or rebootIP phone, gateway, or softphone crashes without sending SIP BYE or CANCELCDR shows call starting normally then continuing for extended period with no mediaMedium โ€” sporadic occurrence depending on endpoint stability
One-way audio scenarioMedia flows in one direction only; one endpoint sends RTP but the other cannot receive or respondAsymmetric RTP; one direction shows zero packets in captureMedium โ€” common with firewall and NAT misconfigurations
Firewall state table overflowFirewall drops RTP session state due to table overflow; SIP session on different port survivesSudden media loss during peak traffic; call remains in signaling stateHigh โ€” affects many calls simultaneously during peak hours
Codec renegotiation failureRe-INVITE for codec change fails on media path but succeeds on signaling pathCall connected with initial codec, then media stops after re-INVITELow โ€” rare but difficult to diagnose
SIP ALG interferenceRouter SIP ALG modifies SDP in ways that break RTP path while keeping SIP signaling functionalCall answers but no RTP flows from the start; stays connected until timeoutMedium โ€” common with consumer-grade routers

How VOS3000 No Media Hangup Works

The VOS3000 no media hangup feature provides an automatic mechanism to detect and terminate ghost calls. When enabled, VOS3000 continuously monitors the RTP media stream for each active call. If no RTP packets are received for the duration specified by the SS_NOMEDIAHANGUPTIME parameter, VOS3000 automatically sends a SIP BYE message to terminate the call and close the session.

The monitoring process works at the media relay level. When VOS3000 operates in Media Proxy mode, all RTP packets pass through the VOS3000 server. The media relay component tracks RTP packet reception for each active call session. If the RTP stream for a call stops โ€” meaning no RTP packets are received on either the caller or callee media port for the configured timeout period โ€” the system considers the call dead and initiates automatic disconnect by sending a SIP BYE to both endpoints.

This Smart detection mechanism is fundamentally different from the SIP session timer. The session timer operates at the SIP signaling layer and detects when SIP re-INVITE or UPDATE refreshes fail. The no media hangup operates at the RTP media layer and detects when voice packets stop flowing, regardless of whether the SIP signaling path is still alive. For details on the session timer mechanism, see our VOS3000 session timer 32-second drop guide.

The Auto-Disconnect Process Step by Step

When VOS3000 detects that no RTP media has been received for a call within the configured timeout, the following sequence occurs:

  1. RTP monitoring: The VOS3000 media relay continuously tracks RTP packet reception for every active call session
  2. Timeout detection: When no RTP packets are received for SS_NOMEDIAHANGUPTIME seconds on a call, the media relay flags the session as dead
  3. BYE generation: VOS3000 generates a SIP BYE request for the affected call and sends it to both the caller and callee endpoints
  4. Session teardown: The SIP dialog is terminated, media relay ports are released, and the call session state is cleaned up
  5. CDR closure: The CDR record is finalized with the disconnect time and appropriate cause code, recording the actual duration the call remained active
VOS3000 No Media Hangup Detection Flow:

1. Call established (SIP 200 OK received and ACKed)
2. RTP media proxy active โ€” packets flowing in both directions
3. RTP stream stops (no packets received from either endpoint)
4. Timer starts: counting seconds since last RTP packet received
5. Timer reaches SS_NOMEDIAHANGUPTIME seconds โ€” call flagged as ghost
6. VOS3000 sends SIP BYE to both endpoints
7. Call session terminated, media ports released, CDR closed

Key Requirement: Media Proxy mode must be active for RTP monitoring.
Direct media bypass mode does NOT support no media hangup detection.

For help configuring Media Proxy mode to support no media hangup detection, refer to the VOS3000 system parameter documentation or contact your system administrator.

Configuring SS_NOMEDIAHANGUPTIME in VOS3000

The SS_NOMEDIAHANGUPTIME parameter is the core configuration for the VOS3000 no media hangup feature. It defines the number of seconds VOS3000 waits without receiving any RTP packets before automatically disconnecting the call. This parameter is configured in the VOS3000 softswitch system parameters, as documented in VOS3000 Manual Section 4.3.5.2.

To configure SS_NOMEDIAHANGUPTIME, follow these steps:

  1. Log in to VOS3000: Access the VOS3000 client application with an administrator account
  2. Navigate to System Parameters: Go to Operation Management > Softswitch Management > Additional Settings > System Parameter
  3. Locate SS_NOMEDIAHANGUPTIME: Search for the parameter name in the system parameter list
  4. Set the timeout value: Enter the desired number of seconds (see configuration values table below)
  5. Save and apply: Save the parameter change โ€” the setting takes effect for new calls; existing calls use the previous value
โš™๏ธ Parameter Value๐Ÿ“ Behavior๐ŸŽฏ Use Caseโš ๏ธ Consideration
0No media hangup disabled โ€” ghost calls never auto-disconnectedWhen relying entirely on SIP session timer for call cleanupGhost calls will persist indefinitely without session timer
30Disconnect after 30 seconds of no RTP mediaAggressive cleanup for high-capacity systems where every slot countsMay disconnect legitimate calls with long silent periods (hold, mute)
60Disconnect after 60 seconds of no RTP mediaBalanced setting for most wholesale VoIP deploymentsGood balance between cleanup speed and legitimate silence tolerance
90Disconnect after 90 seconds of no RTP mediaConservative setting for environments with frequent short silent periodsGhost calls may persist up to 90 seconds before cleanup
120Disconnect after 120 seconds of no RTP mediaVery conservative; maximum tolerance for silent periodsLong ghost call duration before disconnect; wastes more capacity
180+Extended timeout beyond typical recommendationsSpecial scenarios with very long expected silence (intercom systems, paging)Not recommended for general VoIP; ghost calls linger too long
VOS3000 SS_NOMEDIAHANGUPTIME Configuration:

Navigation: Operation Management > Softswitch Management
            > Additional Settings > System Parameter

Parameter:  SS_NOMEDIAHANGUPTIME
Type:       Integer (seconds)
Default:    0 (disabled)
Recommended: 60 seconds for most wholesale deployments

IMPORTANT:
- Value of 0 disables the feature entirely
- Applies only to new calls after the parameter is saved
- Existing calls continue with the previously active setting
- Media Proxy mode MUST be enabled for this feature to function

Setting the Appropriate Timeout

Choosing the right value for SS_NOMEDIAHANGUPTIME requires balancing two competing concerns. A timeout that is too short risks disconnecting legitimate calls where one or both parties are silent for an extended period โ€” for example, during a hold, mute, or a natural pause in conversation. A timeout that is too long allows ghost calls to waste concurrent call capacity and inflate CDR durations before they are finally cleaned up.

The key insight is that RTP packets are normally sent continuously during a VoIP call, even when the parties are silent. This is because most codecs โ€” including G.711, G.729, and G.723 โ€” generate RTP packets containing silence or comfort noise data. Even when both parties are completely silent, RTP packets continue to flow at the codec’s packetization rate (typically every 20ms or 30ms). The only time RTP stops flowing on a legitimate call is when there is a genuine network or endpoint failure.

However, some codecs and configurations implement silence suppression (also called Voice Activity Detection or VAD), which stops sending RTP packets during silent periods. If your deployment uses VAD-enabled codecs, you must set SS_NOMEDIAHANGUPTIME high enough to accommodate the longest expected silence period. For most deployments without VAD, a 60-second timeout provides an excellent balance between rapid ghost call cleanup and tolerance for legitimate call scenarios.

No Media Hangup vs Session Timer: Critical Differences

VOS3000 provides two separate mechanisms for detecting and cleaning up dead calls: the no media hangup feature and the SIP session timer. Understanding the differences between these two mechanisms is essential for proper configuration and avoiding the common confusion between them.

๐Ÿ“Š Aspect๐Ÿ‘ป No Media Hangupโฑ๏ธ Session Timer
Protocol layerRTP media layerSIP signaling layer
What it monitorsRTP packet reception โ€” whether media is flowingSIP re-INVITE/UPDATE refresh โ€” whether signaling session is alive
Detection methodNo RTP packets received for X secondsSIP session refresh fails (re-INVITE timeout)
Trigger conditionMedia path failure while SIP signaling may still be aliveSIP signaling path failure; both signaling and media are dead
Typical timeout30-120 seconds (configurable via SS_NOMEDIAHANGUPTIME)32 seconds default drop after session refresh failure
ParameterSS_NOMEDIAHANGUPTIMESession-Expires header and Min-SE in SIP messages
Catches ghost calls?Yes โ€” detects calls with dead media but live signalingNo โ€” session timer refresh requires signaling to fail; ghost calls have live signaling
Media Proxy required?Yes โ€” must proxy media to monitor RTPNo โ€” operates purely in SIP signaling layer
Best forDetecting ghost calls where media dies but signaling survivesDetecting total signaling failure where both SIP and RTP are dead

The critical takeaway is that the session timer alone cannot catch ghost calls. When a call becomes a ghost โ€” media is dead but SIP signaling is still alive โ€” the session timer refresh succeeds because the SIP path is functional. Only the no media hangup feature can detect this specific condition because it monitors the RTP stream independently of the SIP signaling state. For complete call cleanup, both mechanisms should be configured together. Learn more about the session timer in our VOS3000 session timer 32-second drop guide.

Media Proxy Mode Interaction with No Media Hangup

The VOS3000 no media hangup feature has a critical dependency on Media Proxy mode. Because the detection mechanism works by monitoring RTP packet reception at the media relay level, the media proxy must be active for each call that you want to monitor. If calls are established in direct media bypass mode โ€” where RTP flows directly between endpoints without passing through the VOS3000 server โ€” the no media hangup feature cannot detect ghost calls because the server never sees the RTP packets.

๐Ÿ”ง Media Mode๐Ÿ‘ป No Media Hangup๐Ÿ“ RTP Visibilityโš ๏ธ Notes
Media Proxy (Relay)โœ… Fully functionalAll RTP packets pass through VOS3000; full monitoring capabilityRecommended mode for ghost call detection
Media Bypass (Direct)โŒ Not functionalRTP flows directly between endpoints; VOS3000 cannot monitor packetsGhost calls will NOT be detected in bypass mode
Mixed Modeโšก Partially functionalOnly proxied calls are monitored; bypassed calls are invisibleInconsistent ghost call detection across your traffic

To ensure complete ghost call detection, configure your VOS3000 system to use Media Proxy mode for all calls. This means setting the appropriate media relay configuration for your gateways and ensuring that calls are not falling through to direct media bypass. The tradeoff is slightly higher server resource consumption, as the media relay must process and forward every RTP packet. However, the benefit of automatic ghost call cleanup far outweighs the marginal increase in CPU and bandwidth usage for most deployments.

For guidance on configuring Media Proxy mode and optimizing server resources, see our VOS3000 RTP media guide and VOS3000 system parameters guide. For hands-on assistance, contact us on WhatsApp at +8801911119966.

Detecting Ghost Calls in CDR: Identifying the Patterns

Even with no media hangup configured, you should regularly audit your CDR records to identify ghost call patterns. Ghost calls leave distinctive signatures in CDR data that can be detected through analysis. Early detection of ghost call patterns helps you identify network issues, endpoint problems, and configuration gaps before they cause significant billing disputes.

๐Ÿ” CDR Pattern๐Ÿ‘ป Indicates๐Ÿ“Š Typical Valuesโœ… Action
Very long duration with zero billed amountGhost call that was eventually cleaned up by no media hangupDuration: 60-300 seconds; Billed: $0.00Verify no media hangup is working; check if timeout is appropriate
Unusually long duration with near-zero billed amountGhost call with minimal media before timeoutDuration: hundreds of seconds; Billed: fractions of a centReduce SS_NOMEDIAHANGUPTIME if too many calls affected
Multiple calls from same endpoint with identical long durationsSystematic endpoint or network issue causing repeated ghost callsDuration: matches SS_NOMEDIAHANGUPTIME value consistentlyInvestigate the specific endpoint; check NAT, firewall, and network path
Calls that end exactly at the no media hangup timeoutNo media hangup is actively cleaning up ghost callsDuration: matches SS_NOMEDIAHANGUPTIME + initial media periodFeature is working correctly; investigate root cause of media loss
Disproportionate ACD (Average Call Duration) for specific routesRoute-level network issues causing ghost callsACD significantly higher than expected for the destinationCheck the vendor/gateway for that route; test media path quality
Spike in concurrent call count without corresponding traffic increaseAccumulating ghost calls during a network eventConcurrent calls near license limit; CDR shows many long-duration callsVerify no media hangup is enabled; check Media Proxy mode is active

Using Current Call Monitor for Real-Time Detection

VOS3000 provides a real-time Current Call monitor that shows all active calls on the system. During a network event, you can use the Current Call monitor to identify ghost calls in real time:

  1. Open Current Call: Navigate to Operation Management > Call Management > Current Call
  2. Sort by duration: Click the duration column to sort calls from longest to shortest
  3. Identify anomalies: Calls with unusually long durations, especially from the same endpoint or gateway, are likely ghost calls
  4. Check media status: If available, observe whether the media relay shows active RTP for each call
  5. Manual disconnect: You can manually disconnect suspected ghost calls from the Current Call interface

Regular monitoring of the Current Call screen helps you identify ghost call patterns early and confirm that your SS_NOMEDIAHANGUPTIME configuration is working effectively.

Different call scenarios have different tolerance levels for silence periods, and the SS_NOMEDIAHANGUPTIME value should be set according to the most sensitive call type in your deployment. The following table provides recommended timeout values based on common VoIP call types and their expected media behavior.

๐Ÿ“ž Call Typeโฑ๏ธ Recommended Timeout๐Ÿ’ก Reasoningโš ๏ธ Risk of Too Short
Wholesale termination30-60 secondsHigh call volume; every slot matters; minimal silence expectedBrief holds during IVR transfer could be disconnected
Retail VoIP60-90 secondsEnd users may mute or hold; need more tolerance for natural silenceUsers on hold may be disconnected unexpectedly
Call center / IVR90-120 secondsIVR menus and queue hold times create extended silence periodsCallers in queue may be dropped while waiting for agent
SIP trunking60 secondsPBX trunk connections; moderate silence tolerance neededPBX hold music should generate RTP; silence may indicate real problem
VAD-enabled endpoints120-180 secondsVoice Activity Detection suppresses RTP during silence; needs longer timeoutNormal silent conversation gaps will trigger disconnect
Emergency services120+ seconds (or disable)Never disconnect emergency calls; silence may be critical situationDisconnecting emergency calls is dangerous and may violate regulations

If your VOS3000 deployment handles multiple call types, set SS_NOMEDIAHANGUPTIME to accommodate the most sensitive call type that requires the longest silence tolerance. Alternatively, consider separating different call types onto different VOS3000 instances or prefixes with different configurations. For guidance on optimizing timeout settings for your specific traffic mix, contact us on WhatsApp at +8801911119966.

Use Case: Preventing Billing Disputes from Ghost Calls

One of the most impactful applications of the VOS3000 no media hangup feature is preventing billing disputes. Consider a scenario common in wholesale VoIP: a carrier routes 10,000 calls per day through a vendor gateway. During a 2-hour network instability event, 200 calls lose their RTP media path but remain connected in SIP signaling. Without no media hangup, these 200 ghost calls persist until the endpoints time out or the session expires โ€” potentially lasting 4-6 hours each.

The CDR records show 200 calls with durations of 4-6 hours each. When the billing system calculates charges based on these CDR durations, the customer is billed for 800-1200 hours of call time that had no actual voice communication. When the customer reviews their invoice and CDR records, they find hundreds of calls with extremely long durations and dispute the entire batch of charges. The dispute resolution process consumes significant staff time, and the carrier often has to issue credits to maintain the business relationship.

With VOS3000 no media hangup configured with SS_NOMEDIAHANGUPTIME set to 60 seconds, each ghost call is detected and terminated within 60 seconds of media loss. The 200 ghost calls generate CDR records showing durations of approximately 60 seconds instead of 4-6 hours. The total billed time is reduced from 800-1200 hours to approximately 3.3 hours, and the customer’s CDR shows reasonable call durations that match actual usage. Billing disputes are minimized, and the carrier’s revenue integrity is maintained.

For a complete understanding of VOS3000 billing and how CDR records are generated, see our VOS3000 billing system guide.

Use Case: Freeing Up Concurrent Call Capacity During Network Issues

Concurrent call capacity is a finite and valuable resource in any VOS3000 deployment. Your VOS3000 license determines the maximum number of simultaneous calls the system can handle, and every ghost call consumes one of these precious slots. During network instability events, ghost calls can accumulate rapidly, potentially exhausting your concurrent call capacity and blocking legitimate traffic.

Consider a VOS3000 system licensed for 2,000 concurrent calls during normal operation. The system typically handles 1,500-1,800 concurrent calls during peak hours, leaving 200-500 slots of headroom. A network event causes media loss on 500 calls, but SIP signaling survives on 400 of them. Without no media hangup, those 400 ghost calls remain connected indefinitely, reducing available capacity to 1,600 slots. When peak hour traffic arrives, the system hits the 2,000-call license limit with 400 ghost calls consuming capacity, and legitimate calls start failing with 503 Service Unavailable.

With VOS3000 no media hangup enabled, those 400 ghost calls are automatically terminated within 60 seconds of media loss. The 400 call slots are immediately freed up and available for legitimate traffic. The system maintains its full capacity for real calls, and the network event passes without any impact on call completion rates. This Smart automatic cleanup ensures that your concurrent call capacity is always available for genuine traffic, not wasted on zombie sessions.

Troubleshooting: Legitimate Calls Being Disconnected

The most common problem encountered with VOS3000 no media hangup is legitimate calls being incorrectly disconnected. This happens when the SS_NOMEDIAHANGUPTIME value is set too low for the actual silence patterns in your call traffic. When legitimate calls are disconnected, users experience unexpected call drops, and the CDR shows the disconnect reason as “no media” rather than a normal call termination.

Symptoms of Incorrect Disconnection

  • Users report unexpected call drops: Callers complain that calls are disconnected during normal conversation, especially during pauses or hold periods
  • CDR shows no media disconnect code: The CDR disconnect reason indicates no media timeout rather than a normal BYE from an endpoint
  • Drops correlate with silence periods: Call drops tend to happen during IVR menus, hold periods, or natural conversation pauses
  • Issue affects specific call types: Only certain routes or endpoints are affected, typically those with VAD enabled or those that generate silence during normal operation

Resolving Incorrect Disconnection

  1. Increase SS_NOMEDIAHANGUPTIME: The most direct solution is to increase the timeout value. If calls are being disconnected at 30 seconds, try 60 seconds. If 60 seconds is too aggressive, try 90 seconds
  2. Check for VAD-enabled endpoints: If any endpoints use Voice Activity Detection, RTP stops during silence. Either disable VAD on those endpoints or increase the timeout to accommodate silence periods
  3. Verify Media Proxy is correctly configured: In rare cases, Media Proxy misconfiguration can cause the server to miss RTP packets that are actually flowing. Verify that the media relay is processing packets correctly using packet capture
  4. Analyze specific affected calls: Use SIP trace and RTP capture to examine the calls being disconnected. Confirm that RTP truly stops before the timeout, or whether the monitoring is incorrectly reporting no media
  5. Consider per-route configuration: If only certain routes or endpoints are affected, consider whether you can isolate those calls and apply different settings

For help diagnosing and resolving no media hangup disconnection issues, see our VOS3000 audio troubleshooting guide or contact us on WhatsApp at +8801911119966.

Configuration and Testing Checklist (VOS3000 no media hangup)

Use this checklist to ensure your VOS3000 no media hangup configuration is complete and working correctly before relying on it in production. Each step should be verified and documented.

โœ… Step๐Ÿ“‹ Action๐Ÿ“ Detailsโš ๏ธ Important
1Verify Media Proxy mode is activeCheck that calls are being proxied, not bypassed, in the media relay configurationNo media hangup does NOT work in bypass mode
2Set SS_NOMEDIAHANGUPTIMENavigate to Softswitch Management > System Parameter and set the timeout value in secondsStart with 60 seconds; adjust based on your call types
3Test with a legitimate callPlace a normal test call and verify it stays connected during normal conversationEnsure the timeout does not affect normal calls
4Test ghost call detectionSimulate a ghost call by establishing a call and then blocking RTP on one endpointCall should disconnect within SS_NOMEDIAHANGUPTIME seconds of RTP loss
5Verify CDR recordsCheck that CDR shows correct disconnect reason for the auto-disconnected callCDR should show no media timeout as the disconnect cause
6Test with hold/mute scenarioPlace a call, put one side on hold, and verify the call stays connectedHold music should generate RTP; if not, timeout may trigger
7Monitor Current Call during peakWatch the Current Call screen during peak hours for ghost call accumulationConcurrent call count should not spike abnormally during network events
8Audit CDR for ghost call patternsAfter 24 hours, review CDR for calls matching ghost call patterns (long duration, zero billing)Ghost call patterns should be eliminated or significantly reduced
9Configure session timer as backupEnsure SIP session timer is also configured for total signaling failure scenariosNo media hangup + session timer = complete call cleanup coverage
10Document configurationRecord SS_NOMEDIAHANGUPTIME value, Media Proxy mode, and session timer settingsEssential for future troubleshooting and configuration audits
VOS3000 No Media Hangup Configuration Summary:

Step 1: Verify Media Proxy mode is active for all call paths
Step 2: Set SS_NOMEDIAHANGUPTIME = 60 (recommended starting value)
Step 3: Save system parameter changes
Step 4: Test with legitimate call โ€” verify no false disconnects
Step 5: Simulate ghost call โ€” verify auto-disconnect works
Step 6: Check CDR records for correct disconnect reason
Step 7: Monitor Current Call during peak hours
Step 8: Audit CDR after 24 hours for ghost call patterns
Step 9: Configure SIP session timer as additional safety net
Step 10: Document all settings for future reference

Both no media hangup AND session timer should be configured
for complete protection against dead calls.

FAQ: VOS3000 No Media Hangup

1. What is no media hangup in VOS3000?

No media hangup is a VOS3000 feature that automatically disconnects calls when the RTP media stream stops flowing. It monitors RTP packet reception for each active call through the media relay. When no RTP packets are received for the duration specified by the SS_NOMEDIAHANGUPTIME parameter, VOS3000 sends a SIP BYE to terminate the call. This Smart mechanism prevents ghost calls โ€” calls that remain connected in SIP signaling but have no active voice media โ€” from wasting concurrent call capacity and corrupting CDR billing records. The feature is documented in VOS3000 Manual Section 4.3.5.2 and requires Media Proxy mode to be active for RTP monitoring.

2. What is the SS_NOMEDIAHANGUPTIME parameter?

SS_NOMEDIAHANGUPTIME is a VOS3000 softswitch system parameter that defines the number of seconds the system waits without receiving any RTP packets before automatically disconnecting a call. The parameter is configured in Operation Management > Softswitch Management > Additional Settings > System Parameter. A value of 0 disables the feature entirely. Common production values range from 30 to 120 seconds, with 60 seconds being the recommended starting point for most wholesale VoIP deployments. The parameter only takes effect for new calls after it is saved; existing calls continue with the previously active value.

3. How do ghost calls affect VoIP billing?

Ghost calls have a direct and damaging impact on VoIP billing accuracy. When a call becomes a ghost โ€” SIP signaling remains connected but RTP media stops โ€” the CDR timer continues to run. The CDR records the full duration from call answer to eventual disconnect, including potentially hours of dead air time. The billing system calculates charges based on these inflated CDR durations, resulting in customers being billed for time when no voice communication was actually happening.

This leads to billing disputes, credit requests, and damaged business relationships. The VOS3000 no media hangup feature addresses this by automatically terminating ghost calls within the configured timeout, keeping CDR durations accurate and proportional to actual media activity. For more on billing accuracy, see our VOS3000 billing system guide.

4. What is the difference between no media hangup and session timer?

No media hangup and the SIP session timer are two distinct call cleanup mechanisms in VOS3000 that operate at different protocol layers and detect different failure conditions. No media hangup operates at the RTP media layer โ€” it monitors whether voice packets are flowing and disconnects calls when media stops. The session timer operates at the SIP signaling layer โ€” it uses periodic SIP re-INVITE or UPDATE messages to verify that the SIP signaling path is alive and disconnects calls when the session refresh fails. The critical difference is that ghost calls typically have live SIP signaling but dead RTP media.

The session timer cannot detect ghost calls because the SIP refresh succeeds, while no media hangup can detect them because it monitors the media stream independently. Both mechanisms should be configured together for complete call cleanup coverage.

5. Why are legitimate calls being disconnected by no media hangup?

Legitimate calls are typically disconnected by the no media hangup feature when the SS_NOMEDIAHANGUPTIME value is set too short for the actual silence patterns in your call traffic. The most common cause is endpoints using Voice Activity Detection (VAD), which stops sending RTP packets during silent periods. If VAD is enabled and a caller pauses for longer than SS_NOMEDIAHANGUPTIME seconds, the system interprets the silence as a dead call and disconnects it.

Other causes include long IVR menu pauses, extended hold times without hold music generating RTP, and network jitter causing temporary RTP gaps. The solution is to increase SS_NOMEDIAHANGUPTIME to a value that accommodates the longest expected legitimate silence period, disable VAD on endpoints, or ensure that hold music and IVR prompts generate continuous RTP output.

6. How do I detect ghost calls in CDR records?

Ghost calls leave distinctive patterns in CDR records that can be identified through analysis. The most obvious indicator is a call with an unusually long duration but a zero or near-zero billed amount โ€” this suggests the call had no actual media flowing. Other patterns include: multiple calls from the same endpoint with identical durations matching the SS_NOMEDIAHANGUPTIME value; calls that end exactly at the no media hangup timeout plus the initial media period; and disproportionate Average Call Duration (ACD) for specific routes compared to expected values. To detect ghost calls systematically, sort your CDR by duration in descending order and review the top results.

Look for calls that are significantly longer than the typical ACD for their destination, especially if they cluster around specific endpoints, gateways, or time periods. For monitoring best practices, see our VOS3000 system parameters guide.

7. Does no media hangup work with media bypass mode in VOS3000?

No, the VOS3000 no media hangup feature does not work when calls are in media bypass (direct) mode. The feature relies on the media relay component to monitor RTP packet reception for each active call. In bypass mode, RTP media flows directly between the two endpoints without passing through the VOS3000 server, so the system has no visibility into whether packets are being exchanged. Without access to the RTP stream, the no media hangup timer cannot detect when media stops flowing.

For this reason, you must configure Media Proxy (relay) mode on your VOS3000 gateways and trunks if you want ghost call detection. In a mixed-mode deployment where some calls use proxy and others use bypass, only the proxied calls benefit from no media hangup protection, while bypassed calls remain vulnerable to ghost call accumulation.

Conclusion – VOS3000 no media hangup

Ghost calls are a persistent threat to VoIP operations, silently consuming concurrent call capacity, inflating CDR durations, and generating billing disputes that erode customer confidence. The VOS3000 no media hangup feature, configured through the SS_NOMEDIAHANGUPTIME system parameter, provides a Smart and effective solution by automatically detecting and terminating calls when RTP media stops flowing.

Key takeaways from this guide:

  • Ghost calls occur when SIP signaling survives but RTP media dies โ€” they are invisible to both parties and persist until explicitly terminated
  • SS_NOMEDIAHANGUPTIME controls the auto-disconnect timeout โ€” set it to 60 seconds for most wholesale deployments; 0 disables the feature
  • Media Proxy mode is required โ€” the feature only works when VOS3000 is proxying RTP media, not in bypass mode
  • No media hangup and session timer serve different purposes โ€” configure both for complete call cleanup coverage
  • Choose your timeout carefully โ€” too short disconnects legitimate calls; too long wastes capacity on ghost calls
  • Monitor CDR patterns regularly โ€” ghost call signatures in CDR data reveal network issues before they cause major problems

By implementing VOS3000 no media hangup with the appropriate timeout for your traffic patterns, you can eliminate ghost calls, protect billing accuracy, and ensure that your concurrent call capacity is always available for genuine voice traffic. For professional VOS3000 configuration and support, visit VOS3000 downloads or contact us on WhatsApp at +8801911119966.


๐Ÿ“ž Need Professional VOS3000 Setup Support?

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

๐Ÿ“ฑ WhatsApp: +8801911119966
๐ŸŒ Website: www.vos3000.com
๐ŸŒ Blog: multahost.com/blog
๐Ÿ“ฅ Downloads: VOS3000 Downloads


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