VOS3000 CDR File Rotation, VOS3000 Real-Time CDR Forwarding, VOS3000 CDR Query Blackout, VOS3000 CDR Query Date Range, VOS3000 CDR Text File Export, VOS3000 CDR Pipe Format, VOS3000 CDR Billing Mode Codes, VOS3000 CDR End Direction Critical

VOS3000 CDR End Direction Critical Call Termination Party Detection

VOS3000 CDR End Direction Critical Call Termination Party Detection

๐Ÿ“ž Knowing who hung up the call is not just a curiosity โ€” it is a critical data point that affects billing disputes, quality analysis, fraud detection, and network performance optimization. The VOS3000 CDR end direction field records exactly which party initiated the call termination: the caller (0), the callee (1), or the VOS3000 server itself (2). This three-code system, documented in the official VOS3000 manual ยง4.4 (page 242), provides the definitive answer to “who ended this call?” โ€” and that answer has far-reaching implications for your VoIP business. ๐Ÿ”

โš™๏ธ Consider the billing dispute scenario: A customer claims they were overcharged because “the call dropped after only a few seconds.” Without the endDirection field, you have no way to prove whether the customer hung up normally, the far end hung up, or the server terminated the call due to a timeout or balance exhaustion. With endDirection = 2 (server), you can explain that the server terminated the call because the prepaid balance was depleted โ€” resolving the dispute with evidence. Without it, you are relying on guesswork. ๐Ÿ“Š

๐ŸŽฏ This guide provides a comprehensive reference for the VOS3000 CDR end direction field, covering all three codes (0, 1, 2), their meanings, how they interact with other CDR fields like endReason and billingMode, and practical analysis techniques for using end direction data in billing, quality monitoring, and security applications. All definitions are sourced from the official VOS3000 2.1.8.0/2.1.9.07 English manual ยง4.4 (page 242). ๐Ÿ“˜

๐Ÿ” What Is VOS3000 CDR End Direction?

๐Ÿ“‹ The VOS3000 CDR end direction (also called “hangup side” in the manual) is Field 7 in the pipe-delimited CDR format. It records which party initiated the termination of the call by sending the SIP BYE message, H.323 EndSessionCommand, or equivalent call teardown signal. This is not about which party originated the call โ€” it is specifically about which party ended it.

๐Ÿ“ CDR field location: Position 7 in the pipe-delimited CDR format, between the endReason (Field 6) and callerGatewayId (Field 8) fields, as documented in the VOS3000 manual ยง4.4.

๐Ÿ“‹ Official Manual Definition

๐Ÿ“– The VOS3000 2.1.8.0/2.1.9.07 English manual ยง4.4 (page 242) defines the endDirection field as:

“endDirection โ€” Hangup side๏ผˆ0-caller๏ผŒ1-callee๏ผŒ2-server๏ผ‰”

๐Ÿ“ This is the complete and official definition. The three possible values and their meanings are:

CodePartyMeaningSIP Signal
0๐Ÿ”” CallerThe calling party initiated the hangupBYE from caller side
1๐Ÿ“ž CalleeThe called party initiated the hangupBYE from callee side
2๐Ÿ–ฅ๏ธ ServerThe VOS3000 server initiated the hangupBYE generated by VOS3000

๐Ÿ“Š End Direction Code 0: Caller Hangup

๐Ÿ”” When the VOS3000 CDR end direction is 0, it means the calling party initiated the call termination. In SIP terms, the BYE message originated from the caller’s side of the call leg. This is the most common end direction for normal completed calls โ€” the person who made the call decides they are done talking and hangs up.

AttributeDetail
๐Ÿ“Œ Code0
๐Ÿ“ PartyCaller (calling party)
๐Ÿ”„ Typical ScenarioNormal call completion โ€” caller hangs up after conversation
๐Ÿ“Š Expected Proportion50โ€“80% of connected calls in most deployments

๐Ÿ“‹ Analysis Implications of Caller Hangup

๐Ÿ’ก What caller hangup tells you: When endDirection = 0, the call followed a normal pattern โ€” the calling party placed the call, the conversation took place, and the caller ended it when finished. This is the expected behavior for the majority of outbound calls. However, if you notice an unusually high percentage of caller hangups with very short hold times (under 3 seconds), it may indicate that callers are reaching the wrong number or encountering audio problems and hanging up immediately.

๐Ÿ“Š Quality correlation: Pair endDirection = 0 with short holdTime values to identify potential audio quality issues. If callers consistently hang up within the first few seconds, there may be a one-way audio problem or incorrect number routing. Cross-reference with the endReason codes to get the full picture โ€” a normal SIP 200 OK with endDirection = 0 and holdTime under 2000ms suggests a quick hangup after audio issues rather than a failed call.

๐Ÿ“Š End Direction Code 1: Callee Hangup

๐Ÿ“ž When the VOS3000 CDR end direction is 1, it means the called party initiated the call termination. The BYE message came from the callee’s side. This typically happens when the person who received the call decides to end the conversation.

AttributeDetail
๐Ÿ“Œ Code1
๐Ÿ“ PartyCallee (called party)
๐Ÿ”„ Typical ScenarioCalled party hangs up after conversation ends
๐Ÿ“Š Expected Proportion15โ€“40% of connected calls in most deployments

๐Ÿ“‹ Analysis Implications of Callee Hangup

๐Ÿ’ก What callee hangup tells you: An endDirection of 1 is perfectly normal for many call scenarios โ€” the called party simply ends the conversation. However, a high proportion of callee hangups, especially combined with short hold times, may indicate that the called parties are not expecting the call (possible spam or unsolicited traffic), or that the audio from the caller side is not reaching the callee properly.

๐Ÿ” Wholesale traffic quality indicator: In wholesale VoIP operations, monitoring the ratio of callee hangups to caller hangups on specific routes helps assess traffic quality. A route with a high percentage of callee hangups and short durations may indicate that the terminating carrier’s end users are rejecting or quickly ending calls โ€” a sign of potential CLI (Caller Line Identification) issues or unwanted traffic. This data supports decisions about route optimization and carrier selection.

๐Ÿ“Š End Direction Code 2: Server Hangup

๐Ÿ–ฅ๏ธ When the VOS3000 CDR end direction is 2, it means the VOS3000 server itself initiated the call termination. This is the most operationally significant of the three codes, because it indicates the softswitch actively intervened to end the call โ€” and the reasons for that intervention directly impact billing, customer experience, and system health. ๐Ÿšจ

AttributeDetail
๐Ÿ“Œ Code2
๐Ÿ“ PartyServer (VOS3000 softswitch)
๐Ÿ”„ Typical ScenarioServer-initiated call termination for policy, timeout, or balance reasons
๐Ÿ“Š Expected Proportion5โ€“20% of connected calls, depending on prepaid ratio

๐Ÿ“‹ When Does Server Hangup Occur?

๐Ÿ–ฅ๏ธ There are several important scenarios where VOS3000 terminates a call from the server side, each with different operational implications:

ScenarioDescriptionEnd ReasonImpact
๐Ÿ’ฐ Balance exhaustionPrepaid account runs out of funds during active callVarious (may show session timeout code)Customer may dispute charges
โฑ๏ธ Session timer expirySIP session timer expires without successful re-INVITE refresh200 (normal) or 408Call duration capped by timer
๐Ÿ”ง Administrative disconnectOperator manually disconnects the call via VOS3000 client200Immediate call termination
๐Ÿ“ก No-media timeoutRTP media stream stops flowing for the configured timeout periodVariousDetects dead calls consuming resources
๐Ÿ›ก๏ธ Maximum duration limitCall exceeds the configured maximum call duration200Policy-based call length cap
๐Ÿ”„ Gateway failover cleanupServer terminates call during gateway switching or failover process503 or other errorCall may be re-routed

๐Ÿ’ก Recording server hangups: Whether CDRs for server-initiated hangups are recorded depends on the SERVER_BILLING_RECORD_SERVER_HANG_UP parameter. When this parameter is On, VOS3000 generates CDR records even when the server initiates the hangup, providing a complete audit trail of all call terminations. When Off, server-initiated hangups may not generate CDR records, creating gaps in your billing and operational data. For detailed configuration guidance, see our server hangup CDR recording guide.

๐Ÿ“‹ End Direction and Billing Dispute Resolution

๐Ÿ’ฐ The VOS3000 CDR end direction field is one of the most powerful tools for resolving billing disputes. When a customer challenges a charge, the endDirection code provides objective evidence of what happened during the call:

Dispute ClaimEnd DirectionResolution
“The call dropped after a few seconds”0 (caller hangup)โœ… The caller (customer) hung up normally โ€” not a dropped call
“I was disconnected unexpectedly”2 (server hangup)โš ๏ธ Server terminated โ€” investigate balance exhaustion or session timeout
“The call was much shorter than billed”1 (callee hangup)โœ… The called party hung up โ€” duration matches CDR holdTime
“I never made this call”0 (caller hangup) with specific callerIp๐Ÿ” Verify the callerIp matches the customer’s registered device

๐Ÿ“Š Evidence chain: For maximum dispute resolution effectiveness, combine the endDirection field with other CDR fields. The endReason code tells you why the call ended, the holdTime tells you how long the conversation lasted, the callerIp confirms where the call originated, and the endDirection tells you who terminated the call. Together, these four fields create an unambiguous evidence chain that resolves most billing disputes. For detailed CDR analysis methodology, see our CDR billing discrepancy guide.

๐Ÿ“Š End Direction and Call Quality Analysis

๐Ÿ“ˆ Analyzing end direction patterns across your traffic reveals important quality trends that are not visible from ASR and ACD metrics alone. Here are the key analysis patterns to monitor:

๐Ÿ“‹ End Direction Distribution Analysis

PatternEnd Direction MixIndicatesAction
โœ… Normal distribution60% caller, 30% callee, 10% serverHealthy traffic with normal call patternsNo action needed โ€” continue monitoring
โš ๏ธ High server hangupServer hangup over 25%Session timeouts, balance exhaustion, or system issuesCheck session timer and prepaid balance settings
๐Ÿ” Short callee hangupCallee hangup with holdTime under 5sCalled parties rejecting calls โ€” possible CLI or spam issueReview caller ID presentation and traffic source
๐Ÿšจ Short caller hangupCaller hangup with holdTime under 3sOne-way audio or wrong number โ€” callers hanging up immediatelyCheck audio quality on affected routes

๐Ÿ“‹ End Direction by Gateway Analysis

๐Ÿ“ก Segmenting end direction data by gateway (using the callerGatewayId and calleeGatewayId fields) reveals gateway-specific quality issues. A gateway that shows an unusually high percentage of server-initiated hangups (endDirection = 2) may have connectivity problems causing session timer expirations. A gateway with a high proportion of short-duration callee hangups may be routing traffic to low-quality destinations where end users reject the calls. This gateway-level analysis supports data-driven routing decisions and helps you identify which carriers deliver the best call completion quality. For gateway performance monitoring techniques, see our gateway analysis reports guide.

๐Ÿ“‹ End Direction and Session Timer Interaction

โฑ๏ธ One of the most important operational interactions is between the VOS3000 CDR end direction and the SIP session timer system. When session timers are enabled, VOS3000 periodically sends re-INVITE messages to refresh the session. If the re-INVITE fails (the endpoint does not respond), VOS3000 terminates the call โ€” and the endDirection will be 2 (server). This is a common scenario for calls that “mysteriously drop” after a fixed interval.

Session Timer ScenarioEnd DirectionEnd ReasonResolution
โœ… Re-INVITE succeeds0 or 1 (normal)200 OKCall continues until party hangs up
โš ๏ธ Re-INVITE fails (NAT issue)2 (server)408 or timeoutCheck NAT keepalive settings
๐Ÿ”ง No session timer support2 (server)Session expiryConfigure SS_SIP_NO_TIMER_REINVITE_INTERVAL
๐Ÿ’ฐ Prepaid balance depleted2 (server)200 OK (normal clear)Expected behavior for prepaid accounts

๐Ÿ’ก Investigating mysterious drops: If customers report calls dropping at consistent intervals (e.g., always at 30 minutes or 2 hours), check the SIP session timer configuration. The session timer interval, combined with the SS_SIP_SESSION_UPDATE_SEGMENT parameter, determines when VOS3000 sends re-INVITE refreshes. If the endpoint does not support session timers and SS_SIP_NO_TIMER_REINVITE_INTERVAL is not configured, VOS3000 may terminate the call after the session timer expires โ€” resulting in endDirection = 2.

๐Ÿ›ก๏ธ Common End Direction Analysis Problems and Solutions

โŒ Problem 1: Excessive Server Hangups (endDirection = 2)

๐Ÿ” Symptom: A high percentage of CDRs show endDirection = 2, indicating the server is terminating many calls.

๐Ÿ’ก Cause: Multiple factors can cause excessive server hangups: session timer misconfiguration, NAT traversal failures causing re-INVITE timeouts, prepaid accounts frequently running out of balance, or RTP timeout detecting dead media streams.

โœ… Solutions:

  • โฑ๏ธ Review SIP session timer settings โ€” ensure SS_SIP_NO_TIMER_REINVITE_INTERVAL provides a safety net for non-timer endpoints
  • ๐ŸŒ Check NAT keepalive settings โ€” failed re-INVITEs through NAT firewalls are a leading cause of server-initiated hangups
  • ๐Ÿ’ฐ Verify prepaid balance thresholds โ€” the mid-call balance warning should alert users before their balance is depleted
  • ๐Ÿ“ก Monitor RTP timeout settings that may be too aggressive for legitimate silent periods in calls

โŒ Problem 2: Billing Disputes Where Customer Claims Call Dropped

๐Ÿ” Symptom: Customer disputes a charge, claiming the call dropped unexpectedly, but the CDR shows endDirection = 0 (caller hangup) with a substantial holdTime.

๐Ÿ’ก Cause: The customer may have accidentally ended the call, or their SIP device may have sent a BYE due to a local issue (not a server-side drop). The CDR end direction provides the objective evidence.

โœ… Solutions:

  • ๐Ÿ“‹ Present the endDirection = 0 record to the customer as evidence that their device initiated the hangup
  • ๐Ÿ” Cross-reference with callerIp to confirm the call originated from the customer’s registered device
  • ๐Ÿ“Š Compare the holdTime with the customer’s claim about call duration
  • ๐Ÿ“ž For endDirection = 2 cases, explain the server termination reason (balance exhaustion, session timeout, etc.)

โŒ Problem 3: Short Callee Hangups Indicating Traffic Quality Issues

๐Ÿ” Symptom: High volume of endDirection = 1 records with very short holdTime values on a specific route or gateway.

๐Ÿ’ก Cause: The called parties are answering and immediately hanging up. This can indicate wrong-number calls, CLI (Caller Line Identification) not being presented correctly, or the traffic being perceived as spam by the called parties.

โœ… Solutions:

  • ๐Ÿ“ž Verify that the caller ID being presented to the called party is correct and recognizable
  • ๐Ÿ”ง Check the caller ID management configuration for the affected mapping gateway
  • ๐Ÿ“Š Analyze the geographic distribution of short callee hangups to identify specific regions or carriers with quality issues
  • ๐Ÿ”„ Consider routing adjustments to avoid low-quality termination carriers

๐Ÿ’ก End Direction Best Practices

๐ŸŽฏ Follow these best practices to maximize the value of VOS3000 CDR end direction data in your operations:

Best PracticeRecommendationReason
๐Ÿ“‹ Always enable server hangup CDR recordingSet SERVER_BILLING_RECORD_SERVER_HANG_UP = On๐Ÿ” Complete audit trail of all call terminations
๐Ÿ“Š Monitor end direction distribution weeklyTrack % of codes 0, 1, 2 across all traffic๐Ÿ“ˆ Early detection of quality and configuration issues
๐Ÿ’ฐ Use end direction in billing dispute workflowsInclude endDirection in dispute resolution SOP๐Ÿ›ก๏ธ Objective evidence resolves disputes faster
๐Ÿ“ก Segment by gateway for quality analysisAnalyze end direction per routing gateway๐Ÿ”ง Data-driven carrier selection and route optimization
โฑ๏ธ Correlate endDirection = 2 with session timerMatch server hangups to timer expiry patterns๐Ÿ”ง Identifies NAT and timer configuration problems

โ“ Frequently Asked Questions

โ“ What does VOS3000 CDR end direction 2 mean?

๐Ÿ–ฅ๏ธ A VOS3000 CDR end direction of 2 means the VOS3000 server initiated the call termination. This occurs when the softswitch actively ends the call, rather than either endpoint (caller or callee) hanging up. Common reasons include: prepaid account balance exhaustion (the server terminates the call when funds run out), SIP session timer expiry (the server did not receive a successful re-INVITE refresh), administrative disconnect by the operator, maximum call duration limit reached, or RTP media timeout detecting a dead media stream. The endDirection = 2 code is documented in the VOS3000 manual ยง4.4 (page 242) as “server” hangup side.

โ“ How do I determine why a server hangup occurred?

๐Ÿ” To determine the specific reason for a server-initiated hangup (endDirection = 2), cross-reference the endDirection field with the endReason field (Field 6) in the same CDR record. The endReason provides the SIP response code or cause code that explains why the call was terminated. For example, endDirection = 2 with endReason = 200 typically indicates a normal server-initiated clear (such as balance exhaustion or maximum duration). EndDirection = 2 with endReason = 408 indicates a timeout. Combining these two fields gives you the complete picture of who ended the call and why.

โ“ Does endDirection affect billing calculations?

๐Ÿ’ฐ The endDirection field itself does not directly change billing calculations โ€” the holdTime field determines the billable duration regardless of who hung up. However, endDirection has indirect billing implications. When endDirection = 2 (server hangup), the call may have been terminated before the natural conversation end, which can lead to customer disputes. When analyzing billing data, filtering by endDirection helps you understand the nature of your call completions and identify patterns that affect revenue, such as premature server terminations due to balance exhaustion on prepaid accounts.

โ“ Can the endDirection value be incorrect?

๐Ÿ”ง In rare cases, the endDirection may not accurately reflect the true termination party. This can happen when a SIP ALG (Application Layer Gateway) or intermediate proxy modifies the BYE message direction, or when a gateway sends a BYE on behalf of an endpoint (making it appear as a callee hangup when the caller actually hung up). If you suspect endDirection inaccuracy, enable SIP debug tracing to capture the actual BYE message flow and verify which IP address sent the termination signal. Check our SIP debug guide for instructions on capturing and analyzing SIP message traces.

โ“ How is endDirection different from endReason in VOS3000 CDR?

๐Ÿ“‹ The endDirection field (Field 7) tells you who terminated the call โ€” caller (0), callee (1), or server (2). The endReason field (Field 6) tells you why the call was terminated โ€” using SIP response codes (200, 486, 503, etc.) or Q.850 cause codes. These two fields answer different questions and must be analyzed together for the complete picture. For example, endDirection = 0 with endReason = 200 means the caller hung up normally. EndDirection = 2 with endReason = 200 means the server terminated the call normally (likely due to balance exhaustion or duration limit). EndDirection = 1 with endReason = 486 means the callee rejected the call with a busy signal.

โ“ Should I always record CDRs for server-initiated hangups?

๐Ÿ“ Yes, it is strongly recommended to record CDRs for server-initiated hangups by setting SERVER_BILLING_RECORD_SERVER_HANG_UP = On. Without these records, your CDR data has gaps โ€” you lose visibility into calls that the server terminated, which are often the most operationally significant calls (balance exhaustion, session timeouts, administrative actions). These records are essential for billing dispute resolution, quality analysis, and system health monitoring. The zero-duration CDR control parameter (SERVER_BILLING_RECORD_ZERO_HOLD_TIME) serves a different purpose โ€” it controls whether failed call attempts are recorded, while SERVER_BILLING_RECORD_SERVER_HANG_UP specifically addresses server-initiated terminations.

๐Ÿ“ž Need Expert Help with VOS3000 CDR End Direction?

๐Ÿ”ง Understanding and analyzing VOS3000 CDR end direction data is essential for billing accuracy, quality monitoring, and operational intelligence. Whether you are investigating server-initiated hangups, resolving billing disputes, or building a call quality dashboard, expert guidance ensures your analysis is accurate and actionable. ๐Ÿ“Š

๐Ÿ’ฌ WhatsApp: +8801911119966 โ€” Get immediate assistance with VOS3000 CDR end direction analysis, billing dispute resolution, and call quality monitoring. Our team specializes in VOS3000 CDR analytics, billing system optimization, and VoIP quality assurance. ๐Ÿ”ง

๐Ÿ”— Explore related VOS3000 CDR and call quality guides:


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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 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


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


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 2.1.9.07 Release Notes

VOS3000 2.1.9.07 Release Notes โ€“ Complete Important Features Upgrade from 2.1.8.05/2.1.8.0

VOS3000 2.1.9.07 Release Notes โ€“ Complete Important Features Upgrade from 2.1.8.05/2.1.8.0

VOS3000 2.1.8.05 and 2.1.9.07 Version Differences, What is New at VOS3000 2.1.9.07 Version, New Updates of VOS3000 2.1.9.07 version – all contains in this VOS3000 2.1.9.07 Release Notes

This document contains the complete and verified VOS3000 2.1.9.07 Release Notes prepared after a detailed comparison between version 2.1.8.05 and 2.1.9.07 manuals. Every new module, routing logic, billing upgrade, SIP enhancement, security feature and backend architectural improvement has been documented.

For more deep technical tutorials visit: VOS3000 Technical Blog


๐Ÿ†• 1. New Major Sections & Functional Modules

๐Ÿงพ 1.1 Modify CDR (Account Management โ€“ 2.4.7)

  • Post-billing correction of charged amount
  • Manual modification of historical CDR charge
  • Administrative billing adjustment control
  • Permission-based modification access

Purpose: Billing correction without database-level manipulation.

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐ŸŒ 1.2 Geofencing (Operation Management โ€“ 2.5.7)

Replaces older โ€œProhibited Media IPโ€ module.

New Capabilities:

  • IP range definition (start IP + count)
  • Signaling IP checking
  • SDP media IP checking
  • RTP actual IP checking

Mode Selection:

  • Ignore
  • Forbidden
  • Allow

Applied At:

  • Global level
  • Routing gateway
  • Mapping gateway
  • Phone management

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ“˜ 1.3 Functional Scenarios (New Chapter 3)

๐Ÿ”น 3.1 First Usage

Updated wholesale deployment quick-start scenario.

๐Ÿ”น 3.2 Pickup Call Transfer

  • Access code configuration
  • Position definition
  • Association settings
  • Functional logic explanation

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿง  1.4 Function Explanation (New Chapter 4.1)

โฑ Network Routing Quality Reserve Time

  • SS_GATEWAY_QUALITY_RESERVE_SEPARATE
  • SS_GATEWAY_QUALITY_RESERVE_TIME

Enables ASR/ACD time-sliced calculation.

๐Ÿ”„ NAT Keep

UDP keep-alive logic to maintain NAT bindings.

โณ SIP Timer Protocol

Session timer support and related parameters.

๐Ÿ“ก Signaling QoS

  • SS_QOS_SIGNAL
  • SS_QOS_RTP

DSCP control for SIP and RTP packets.

๐Ÿ” Enable Bilateral Reconciliation

Real-time reconciliation between two VOS platforms with deviation alarm. VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ›ก 2. Security & Anti-Fraud Enhancements

๐Ÿšซ 2.1 Dynamic Malicious Call Blacklist Engine

  • Concurrent caller limit detection
  • Malicious frequency limit detection
  • No-answer attack detection
  • Time-window based analysis
  • Auto blacklist expiration
  • Dynamic blocking logic
  • Concurrency limit parameters
  • Malicious call check interval
  • Blacklist expiration timer

๐Ÿ” 2.2 Authentication Security Controls

  • Max authentication retry limit
  • Auto suspend after failure
  • Brute-force mitigation logic

๐Ÿ“ก 3. Real-Time Integration & External Control

๐ŸŒ 3.1 Call State HTTP Reporting

  • HTTP call state reporting
  • Configurable report IP
  • Configurable report port
  • Retry mechanism
  • Retry interval control

๐Ÿ”€ 3.2 External SIP Redirect Server (3xx Support)

  • External routing decision server
  • SIP 3xx redirect integration
  • Selective phone availability

๐Ÿ“ฑ 3.3 Phone Service Layer

  • Phone online/offline reporting
  • Dedicated phone service IP & port
  • Offline phone redirect to gateway
  • Phone state monitoring

๐Ÿ”„ 4. Call Handling & Transfer Enhancements

โ˜Ž 4.1 Advanced Transfer Controls

  • Blind transfer key
  • Attended transfer key
  • Wait-access timeout
  • Remote ring passthrough
  • Transfer cancel key
  • Transfer end key
  • Transfer display customization

๐ŸŽต 4.2 Auxiliary Ring Tone

  • Local ringback tone playback
  • SS_AUXILIARY_RING_TONE_ACTIVATION_DELAY

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ“Š 5. Routing & Gateway Enhancements

๐Ÿ›ฃ 5.1 Real-Time Routing Quality Calculation

  • SS_GATEWAY_QUALITY_CALCULATE

๐Ÿ“ˆ 5.2 Routing Strategy

  • Sort by ASR
  • Sort by lowest rate per second

๐Ÿ” 5.3 Bilateral Reconciliation Controls

  • SERVER_GATEWAY_ROUTE_BILATERAL_RECONCILIATION_LINE
  • SERVER_GATEWAY_BILATERAL_RECONCILIATION_PERIOD

๐Ÿ“ž 5.4 Caller Number Pool

  • Enable caller number pool
  • Concurrency per number
  • Forwarding caller pool
  • Multiplex control

๐Ÿšฆ 5.5 Rate Limiting

New signaling rate control per gateway.

๐Ÿงพ 5.6 SIP Enhancements

  • Stop switching response code
  • Reply address mode selection
  • Request address selection
  • G729 annexb control
  • G723 annexa control
  • Enable timer protocol
  • Enable 100rel
  • Retry-After header support
  • Reason header injection
  • user=phone support
  • Allow Publish
  • Enable local domain name
  • Enable call forward signal
  • SIP OPTIONS online check
  • Ptime adaptive
  • NAT media SDP IP first
  • Invite custom header fields
  • Allow all extra header fields
  • Allow specified extra header fields
  • Support Privacy
  • Recognize call forward signal
  • Replace failed reason mapping
  • Remote ringback mode

๐Ÿ”ข 5.7 LRN Handling

  • Eat prefix length
  • Failure action
  • Routing using number
  • Interstate billing prefix
  • Undetermined billing prefix

๐ŸŽ™ 5.8 H.323 Enhancements

  • Q.931 ProgressIndicator
  • Caller field selection
  • Callee field selection
  • Send CallProceeding immediately
  • Convert Trying to CallProceeding
  • Convert 183(SDP) to Alerting

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ“‚ 6. CDR & Reporting Improvements

๐Ÿงพ 6.1 Enhanced CDR Fields

  • Incoming caller
  • Outgoing caller
  • Connect delay (PDD)
  • Continue duration
  • Billing method
  • Package usage duration
  • Package charges
  • Transparent hangup reason

๐Ÿ“Š 6.2 Reorganized CDR Analysis

  • Mapping Gateway Analysis
  • Routing Gateway Analysis
  • Performance analysis
  • Call analysis
  • Fail analysis
  • Daily call analysis
  • Area analysis
  • Gateway area cross analysis
  • Overall Area analysis

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ’ฐ 7. Billing & Financial Enhancements

๐Ÿ’ณ 7.1 Customer Package (Suite Order System)

  • Subscription packages
  • Effective & expiration control
  • Priority control
  • Free minutes
  • Free amount
  • Minimum consumption
  • Percentage rent
  • Renewal handling rules
  • Failed processing mode selection

๐Ÿ“ 7.2 Billing Precision Controls

  • Billing fee precision
  • Billing unit precision
  • Hold-time precision
  • Overdraft prevention advance time
  • Profit formula logic
  • Gateway route prefix billing
  • Forward prefix billing logic

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ”” 8. Alarm & Monitoring

  • Voice-based notification
  • Passthrough RTP loss rate

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ–ฅ 9. Major Backend Upgrade โ€“ 64 Bit Linux Architecture

Up to version 2.1.8.05 all backend components were based on 32-bit architecture.

Limitations of 32-bit:

  • ~4GB memory ceiling
  • Limited process scalability
  • Lower high-concurrency stability

2.1.9.07 Backend Improvements:

  • Full 64-bit Linux architecture
  • High RAM utilization (32GB / 64GB / 128GB+)
  • Better multi-core CPU usage
  • Improved database caching
  • Higher CPS handling capability
  • Better memory allocation efficiency
  • Improved stability under heavy wholesale traffic
VOS3000 2.1.9.07 sample RPM Installation Files

VOS3000 2.1.9.07 Release Notes is created by AI software from 2 versions user manuals


๐Ÿ“Š Complete Comparison Table โ€“ VOS3000 2.1.8.05 vs 2.1.9.07

Module / FeatureVOS3000 2.1.8.05VOS3000 2.1.9.07
Backend Architecture32-bit Linux64-bit Linux (High RAM Support)
Modify CDR (Post Billing Correction)Not AvailableAvailable
Geofencing (Advanced IP Control)Basic Prohibited Media IPFull Geofencing (Signaling + SDP + RTP)
Dynamic Malicious Call BlacklistNot AvailableAvailable (Auto Detection Engine)
Concurrent Caller DetectionNoYes
No-Answer Attack DetectionNoYes
Authentication Retry ProtectionBasicAdvanced with Auto Suspend
HTTP Call State ReportingNoYes (Real-Time Push API)
External SIP Redirect Server (3xx)NoYes
Phone Service LayerNoYes (Online/Offline Monitoring)
Real-Time Routing Quality CalculationStatic RoutingASR/ACD Real-Time Calculation
Bilateral ReconciliationNoYes
Caller Number PoolNoYes
Signaling Rate LimitingNoYes
SIP Timer ProtocolLimitedEnhanced
SIP 100rel SupportNoYes
Retry-After HeaderNoYes
Reason Header InjectionNoYes
Privacy Header SupportBasicEnhanced
LRN Advanced HandlingLimitedPrefix + Routing Enhancements
H.323 ProgressIndicatorNoYes
Advanced Transfer ControlsBasicBlind + Attended + Cancel + Display
Auxiliary Ring ToneNoYes
Enhanced CDR Fields (PDD, Package Usage)LimitedExpanded Fields
Structured CDR AnalysisBasicAdvanced Gateway & Area Analytics
Customer Package (Suite Order System)NoYes
Billing Precision ControlLimitedAdvanced Precision Parameters
Profit Formula LogicBasicEnhanced
Voice Alarm SupportNoYes
Passthrough RTP Loss StatisticsNoYes
High RAM SupportLimited (~4GB)32GB / 64GB / 128GB+
High CPS StabilityModerateHigh Performance

โ“ FAQ โ€“ VOS3000 2.1.9.07 Release Notes

1. What is the biggest upgrade in VOS3000 2.1.9.07?

The most significant upgrade is the migration to a 64-bit Linux backend architecture, enabling high RAM utilization, improved concurrency handling, and enhanced system stability for wholesale VoIP deployments.

2. Does VOS3000 2.1.9.07 support real-time routing optimization?

Yes. The new real-time routing quality calculation (ASR/ACD based) dynamically sorts gateways based on performance metrics.

3. What is the purpose of the Modify CDR feature?

Modify CDR allows administrators to adjust historical billing charges without directly manipulating the database, improving operational safety and billing correction flexibility.

4. How does the new Geofencing system improve security?

Geofencing validates signaling IP, SDP IP, and actual RTP IP. It can Allow, Ignore, or Block calls based on defined IP ranges, significantly improving fraud prevention.

5. Does this version include anti-fraud protection?

Yes. It introduces a dynamic malicious call blacklist engine with concurrent call detection, frequency monitoring, no-answer attack detection, and automatic blacklist expiration.

6. Can VOS3000 2.1.9.07 integrate with CRM or external billing systems?

Yes. Through HTTP Call State Reporting and External SIP Redirect Server support, real-time integration with CRM, monitoring, and billing platforms is possible.

7. Is bilateral reconciliation supported?

Yes. Two VOS platforms can now perform real-time reconciliation with deviation alarms to prevent financial mismatches.

8. Does 2.1.9.07 improve SIP interoperability?

Yes. It adds support for 100rel, Retry-After, Reason header injection, Privacy handling, advanced NAT processing, and SIP timer protocol enhancements.

9. What billing improvements are included?

The Suite Order System introduces subscription packages, free minutes, minimum consumption, percentage rent billing, and advanced precision control for billing fees and units.

10. Is VOS3000 2.1.9.07 suitable for high-volume wholesale VoIP traffic?

Yes. With 64-bit architecture, improved routing intelligence, anti-fraud engine, and high RAM utilization, it is significantly more stable under heavy traffic compared to 2.1.8.x.



๐Ÿ“ฅ Official Resources

VOS3000 Technical Blog: https://www.vos3000.com/blog/vos3000
Official Blog: https://www.vos3000.com/blog/
Downloads & Manuals: https://www.vos3000.com/downloads.php
Advanced Security Guides: https://multahost.com/blog/


๐Ÿ“ž Contact for VOS3000 Hosting

VOS3000 2.1.9.07 One Time Installation / Hosted (Dedicated Server Only) Available!

๐Ÿ“ฒ WhatsApp: +8801911119966

Direct Link: wa.me/+8801911119966

๐ŸŒ China | Hong Kong | Vietnam | Thailand Servers Available

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