VOS3000 SIP Session Timer: Powerful RFC 4028 Setup Guide
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 — and automatically hangs up dead sessions before they inflate your billing.
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.
This guide walks you through every VOS3000 SIP session timer parameter — from SS_SIP_SESSION_TTL to SS_SIP_NO_TIMER_REINVITE_INTERVAL — with real default values, configuration steps, and best practices to keep your VoIP network clean and profitable.
Table of ContentsVOS3000 SIP Session Timer: Powerful RFC 4028 Setup Guide What Is VOS3000 SIP Session Timer? RFC 4028 Core Concepts for VOS3000 VOS3000 SIP Session Timer Parameters Deep Dive SS_SIP_SESSION_TTL — Detecting SIP Connected Status Interval SS_SIP_SESSION_UPDATE_SEGMENT — Reinvite Interval Divider SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP — Early Hangup Timer SS_SIP_NO_TIMER_REINVITE_INTERVAL — Non-Timer SIP Caller Limit How VOS3000 SIP Session Timer Works — Complete Flow Scenario A: Caller SUPPORTS Session Timer Scenario B: Caller Does NOT Support Session Timer Step-by-Step VOS3000 SIP Session Timer ConfigurationStep 1: Navigate to SIP Parameters Step 2: Configure SS_SIP_SESSION_TTL Step 3: Set SS_SIP_SESSION_UPDATE_SEGMENT Step 4: Configure Early Hangup Step 5: Adjust Non-Timer Caller Limit Complete VOS3000 SIP Session Timer Parameter Reference Common VOS3000 SIP Session Timer Problems and Solutions Problem 1: Calls Dropping Every 5 Minutes Problem 2: Ultra-Long Bills from Zombie Calls Problem 3: Excessive Signaling Overhead VOS3000 SIP Session Timer Best Practices VOS3000 SIP Session Timer and SIP Call Flow Interaction Verifying VOS3000 SIP Session Timer OperationUsing SIP Debug to Confirm Timer Negotiation Frequently Asked Questions What is the default VOS3000 SIP session timer value? How does VOS3000 handle SIP callers that do not support session timers? Can I set SS_SIP_SESSION_UPDATE_SEGMENT to 1? What happens when VOS3000 SIP session timer detects a dead call? Is the VOS3000 SIP session timer compliant with RFC 4028? Should I enable SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP? How does VOS3000 SIP session timer prevent ultra-long bills? Need Expert Help with VOS3000 SIP Session Timer? Related Resources Need Professional VOS3000 Setup Support?
What Is VOS3000 SIP Session Timer?
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.
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:
Periodically sending re-INVITE or UPDATE messages to confirm the call is still alive
Automatically hanging up calls when no confirmation is received
Preventing ultra-long bills caused by zombie sessions
Detecting abnormal network conditions in real time
Location in VOS3000 Client: Navigation → Operation management → Softswitch management → Additional settings → SIP parameter
RFC 4028 Core Concepts for VOS3000
RFC 4028 introduces the Session-Expires header and Min-SE header to SIP. Here’s how they map to VOS3000:
VOS3000 SIP Session Timer Parameters Deep Dive
Let’s examine each parameter in detail using the official VOS3000 2.1.9.07 manual data.
SS_SIP_SESSION_TTL — Detecting SIP Connected Status Interval
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 Parameter NameSS_SIP_SESSION_TTL Default Value600 seconds (10 minutes) UnitSeconds 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.
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.
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.
SS_SIP_SESSION_UPDATE_SEGMENT — Reinvite Interval Divider
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 Parameter NameSS_SIP_SESSION_UPDATE_SEGMENT Default Value2 Range2 – 10 DescriptionSIP Timer reinvite (update) Interval — divides the TTL into segments
Calculation: The actual re-INVITE interval = SS_SIP_SESSION_TTL ÷ SS_SIP_SESSION_UPDATE_SEGMENT
TTL (seconds)SegmentRe-INVITE IntervalUse Case6002300s (5 min) Default — balanced6004150s (2.5 min) More frequent checks6006100s (1.7 min) Unstable networks6001060s (1 min) High overhead18003600s (10 min) Long calls, stable net
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.
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP — Early Hangup Timer
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 Parameter NameSS_SIP_SESSION_TIMEOUT_EARLY_HANGUP Default Value0 seconds (disabled) UnitSeconds DescriptionSIP Timer no reinvite (update) Early Hang up — extra grace period before terminating
When set to 0 (default): VOS3000 hangs up immediately when the session timer expires without confirmation. No grace period is given.
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.
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.
SS_SIP_NO_TIMER_REINVITE_INTERVAL — Non-Timer SIP Caller Limit
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 Parameter NameSS_SIP_NO_TIMER_REINVITE_INTERVAL Default Value7200 seconds (2 hours) UnitSeconds DescriptionIf SIP caller doesn’t support “timer”, softswitch will stop the call when the time is up
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 — once the call duration exceeds this value, VOS3000 forcibly terminates it.
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).
How VOS3000 SIP Session Timer Works — Complete Flow
Understanding the full session timer flow is essential for proper configuration. Here’s exactly what happens during a call:
Scenario A: Caller SUPPORTS Session Timer
Call Established (200 OK)
│
├── VOS3000 starts TTL countdown (SS_SIP_SESSION_TTL = 600s)
│
├── At TTL/Segment = 300s ──► VOS3000 sends re-INVITE/UPDATE
│ ├── 200 OK received → Session confirmed, timer resets
│ └── No response → Retry at next segment
│
├── At TTL = 600s ──► Final check
│ ├── 200 OK received → Session confirmed, timer resets
│ └── No response → Call terminated (BYE sent)
│ └── If EARLY_HANGUP > 0 → Wait X seconds, then BYE
│
└── Cycle repeats for duration of call
Scenario B: Caller Does NOT Support Session Timer
Call Established (200 OK — no Session-Expires header)
│
├── VOS3000 detects no timer support
│
├── No re-INVITE/UPDATE messages sent
│
├── Call continues until…
│ ├── Normal BYE from either party, OR
│ └── Duration exceeds SS_SIP_NO_TIMER_REINVITE_INTERVAL (7200s)
│ └── VOS3000 forcibly terminates call (BYE sent)
│
└── No active session detection possible
Step-by-Step VOS3000 SIP Session Timer Configuration
Follow these steps to configure the VOS3000 SIP session timer parameters:
Step 1: Navigate to SIP Parameters
Log in to VOS3000 Client
Navigate: Operation management → Softswitch management → Additional settings → SIP parameter
Locate the session timer parameters in the parameter list
Step 2: Configure SS_SIP_SESSION_TTL
Deployment TypeRecommended TTLRationale Standard enterprise600s (default) Good balance of detection and overhead High-volume wholesale300s – 600s Faster zombie detection on busy routes Unstable networks180s – 300s Quick detection of dropped calls Premium routes900s – 1800s Less signaling overhead, longer calls OK
Step 3: Set SS_SIP_SESSION_UPDATE_SEGMENT
Choose the segment value based on your network reliability:
Segment ValueTTL=600 IntervalRetry CountBest For2 (default)300s2 attempts Most deployments3200s3 attempts Moderate reliability5120s5 attempts Flaky connections875s8 attempts Very unstable nets
Step 4: Configure Early Hangup
Set SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP based on your tolerance for ghost calls:
0 seconds (default): Immediate hangup — zero tolerance for zombie calls
5-10 seconds: Small grace window for momentary network blips
30+ seconds: Not recommended — defeats the purpose of session timers
Step 5: Adjust Non-Timer Caller Limit
Set SS_SIP_NO_TIMER_REINVITE_INTERVAL based on your risk tolerance:
SettingDurationRisk LevelUse Case7200s (default)2 hours MediumStandard VoIP operations3600s1 hour Low-MediumWholesale termination1800s30 minutes LowHigh-value premium routes900s15 minutes Very LowMaximum protection
Complete VOS3000 SIP Session Timer Parameter Reference
Here’s the full reference table combining all session timer parameters from the official VOS3000 2.1.9.07 manual:
Common VOS3000 SIP Session Timer Problems and Solutions
Even with proper configuration, session timer issues can arise. Here are the most common problems and their fixes:
Problem 1: Calls Dropping Every 5 Minutes
Symptom: Active calls are being terminated at exactly the re-INVITE interval.
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.
Solutions:
Increase SS_SIP_SESSION_TTL to give more time per cycle
Reduce SS_SIP_SESSION_UPDATE_SEGMENT for fewer but longer intervals
Verify the far-end device supports RFC 4028 session timers
Check if the far-end is behind a SIP ALG that drops re-INVITEs — see our SIP debug guide
Problem 2: Ultra-Long Bills from Zombie Calls
Symptom: CDR records show calls lasting hours beyond actual conversation time.
Cause: The SIP caller does not support session timers, and SS_SIP_NO_TIMER_REINVITE_INTERVAL is too high.
Solutions:
Reduce SS_SIP_NO_TIMER_REINVITE_INTERVAL from 7200 to 1800 or lower
Ensure SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP is set to 0 (immediate cleanup)
Monitor CDR records for abnormally long calls — use our CDR billing discrepancy guide
Problem 3: Excessive Signaling Overhead
Symptom: High CPU usage on VOS3000 server, excessive SIP signaling traffic.
Cause: SS_SIP_SESSION_UPDATE_SEGMENT is set too high, causing frequent re-INVITEs.
Solutions:
Reduce SS_SIP_SESSION_UPDATE_SEGMENT to 2 (default) for fewer refresh attempts
Increase SS_SIP_SESSION_TTL to 900 or 1800 for longer cycles
Balance detection speed against signaling load
VOS3000 SIP Session Timer Best Practices
Follow these best practices to get the most from your VOS3000 SIP session timer configuration:
Best PracticeRecommendationReason Start with defaultsTTL=600, Segment=2Proven balance for most deployments Monitor CDRsCheck for abnormally long calls weeklyDetects zombie calls early Lower non-timer limitSet NO_TIMER to 1800–3600Reduces risk from non-RFC 4028 endpoints Test before productionVerify with SIP debug toolsAvoids unexpected call drops Verify endpoint supportCheck Session-Expires in SIP INVITEConfirms timer negotiation works Keep early hangup at 0Unless network is very unstableImmediate cleanup is safer
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.
VOS3000 SIP Session Timer and SIP Call Flow Interaction
The session timer operates within the broader SIP call flow. Understanding how it interacts with other SIP messages is critical:
SIP Call Flow with Session Timer:
Caller ──────────────────── VOS3000 ──────────────────── Called Party
│ │ │
│──── INVITE ────────────────►│──── INVITE ────────────────►│
│ (Session-Expires: 600) │ (Session-Expires: 600) │
│ │ │
│◄─── 200 OK ────────────────│◄─── 200 OK ────────────────│
│ (Session-Expires: 600) │ (Session-Expires: 600) │
│ │ │
│ … call in progress … │
│ │ │
│ ┌─ TTL/Segment timer ──┐ │
│ │ (300s elapsed) │ │
│ └──────────────────────┘ │
│ │ │
│◄─── re-INVITE/UPDATE ──────│──── re-INVITE/UPDATE ──────►│
│ │ │
│──── 200 OK ────────────────►│◄─── 200 OK ────────────────│
│ │ │
│ … timer resets … │
│ │ │
If no 200 OK response: │
│ │──── BYE ────────────────────►│
│◄─── BYE ───────────────────│ │
For a deeper understanding of how session timers fit into the complete SIP call lifecycle, see our comprehensive SIP call flow guide.
Verifying VOS3000 SIP Session Timer Operation
After configuration, verify that session timers are working correctly:
Using SIP Debug to Confirm Timer Negotiation
# 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 <– Session timer negotiated!
Min-SE: 90 <– 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
Need help debugging SIP signaling? Check our SIP debug guide for step-by-step Wireshark capture instructions.
Frequently Asked Questions
What is the default VOS3000 SIP session timer value?
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.
How does VOS3000 handle SIP callers that do not support session timers?
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.
Can I set SS_SIP_SESSION_UPDATE_SEGMENT to 1?
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.
What happens when VOS3000 SIP session timer detects a dead call?
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.
Is the VOS3000 SIP session timer compliant with RFC 4028?
Yes. The VOS3000 SIP session timer implementation follows RFC 4028 — 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.
Should I enable SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP?
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.
How does VOS3000 SIP session timer prevent ultra-long bills?
Ultra-long bills occur when calls remain in “connected” state after the actual conversation has ended — 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.
Need Expert Help with VOS3000 SIP Session Timer?
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.
WhatsApp: +8801911119966 — Get instant support for VOS3000 SIP session timer configuration, RFC 4028 compliance, and VoIP network optimization.
Related Resources
VOS3000 Session Timer — Complete session timer overview
VOS3000 Max Call Duration — Hard call duration limits
VOS3000 SIP Call Flow — Full SIP signaling lifecycle
VOS3000 SIP Debug Guide — Troubleshoot SIP signaling issues
VOS3000 Parameter Description — All parameter reference
VOS3000 CDR Billing Discrepancy — Fix billing accuracy issues
VOS3000 Billing System — Complete billing configuration
VOS3000 System Parameters — System-level configuration
VOS3000 SIP Session — SIP session management
Still have questions about the VOS3000 SIP session timer? Reach out on WhatsApp at +8801911119966 — we provide professional VOS3000 installation, configuration, and support services worldwide.
Need Professional VOS3000 Setup Support?
For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:
WhatsApp: +8801911119966 Website: www.vos3000.com Blog: multahost.com/blog Downloads: VOS3000 Downloads