Las VOS3000 llamadas cortadas son uno de los problemas que mas afectan la experiencia del usuario y la percepcion de calidad de un servicio VoIP. ๐ซ Cuando una llamada se desconecta prematuramente sin que ninguna de las partes haya colgado, el usuario percibe el servicio como poco confiable, independientemente de la calidad de audio durante la llamada. Comprender las causas y aplicar las soluciones correctas es fundamental para mantener la satisfaccion del cliente. ๐ง
En esta guia completa sobre las VOS3000 llamadas cortadas, cubriremos todas las causas posibles de desconexion prematura, desde los temporizadores SIP y RTP hasta los problemas de failover y firewall. Cada seccion incluye tablas de diagnostico, ejemplos y soluciones paso a paso. ๐
Table of Contents
Causas Principales de Llamadas Cortadas ๐
Las VOS3000 llamadas cortadas pueden ser causadas por multiples factores que actuan en diferentes capas del sistema. Identificar la capa donde ocurre el problema es el primer paso para una solucion efectiva. ๐
๐ Causa
Frecuencia
Capa
Sintoma
โฑ๏ธ RTP Timeout
โญโญโญโญโญ Muy alta
Media
Corte despues de silencio
๐ Session Timer
โญโญโญโญ Alta
Senalizacion
Corte a intervalo fijo
๐ฅ Firewall UDP Timeout
โญโญโญโญ Alta
Red
Corte despues de X minutos
๐ Failover/Switch
โญโญโญ Media
Ruteo
Corte con cambio de ruta
๐ Proveedor rechaza
โญโญโญ Media
Terminacion
Corte con codigo SIP
๐ NAT Timeout
โญโญโญโญ Alta
Red
Corte en llamadas largas
RTP Timeout: La Causa Mas Comun โฑ๏ธ
El RTP timeout es la causa mas frecuente de VOS3000 llamadas cortadas. Cuando VOS3000 detecta que no hay flujo RTP en una direccion durante un periodo determinado, asume que la llamada ha perdido conectividad y la desconecta automaticamente. Esto puede ocurrir cuando un dispositivo entra en silencio prolongado o cuando el flujo RTP se interrumpe por problemas de red. ๐
Para solucionar problemas de RTP timeout, configure el parametro RTP timeout en VOS3000 con un valor adecuado (tipicamente 30-60 segundos). Un valor muy bajo causara cortes prematuros durante pausas naturales en la conversacion, mientras que un valor muy alto mantendra llamadas fantasma que consumen recursos. Para informacion sobre RTP, consulte nuestra guia de interrupcion RTP del sistema VOS3000. ๐ง
SIP Session Timer ๐
El SIP Session Timer es un mecanismo que mantiene las sesiones SIP activas mediante re-INVITEs periodicos. Si el Session Timer no se renueva correctamente, se produciran VOS3000 llamadas cortadas a intervalos regulares. Este problema es comun cuando los dispositivos no soportan Session Timer o cuando los re-INVITEs son bloqueados por un firewall. โฑ๏ธ
Para configurar el Session Timer, acceda a los parametros SIP de VOS3000 y defina el intervalo de sesion (tipicamente 1800 segundos). Para informacion sobre sesiones SIP, consulte nuestra guia de sesion SIP del sistema VOS3000. ๐
Firewall UDP Timeout ๐ฅ
Los firewalls que implementan timeouts UDP cortos son una causa muy comun de VOS3000 llamadas cortadas. Los firewalls mantienen una tabla de conexiones UDP activas, y si no ven trafico en una entrada durante un tiempo determinado (tipicamente 5-30 minutos), eliminan la entrada y bloquean los paquetes posteriores. Esto corta la llamada sin que VOS3000 genere un BYE. ๐ฅ
Para solucionar este problema, configure el SIP NAT keepalive en VOS3000 para enviar paquetes periodicos que mantengan las entradas UDP activas en el firewall. El intervalo de keepalive debe ser menor que el timeout UDP del firewall. Para informacion sobre NAT, consulte nuestra guia de NAT del sistema VOS3000. ๐
Failover y Cambio de Ruta ๐
El failover agresivo puede causar VOS3000 llamadas cortadas cuando el sistema detecta degradacion en la ruta actual y conmuta a una ruta alternativa. Si la conmutacion no se realiza correctamente, la llamada existente se corta. Para informacion sobre failover, consulte nuestra guia de failover del sistema VOS3000. ๐
Para configurar el failover sin afectar llamadas en curso, ajuste los parametros de switch limit y aggressive failover en VOS3000. El switch limit define el umbral de llamadas fallidas antes de cambiar de ruta, mientras que el aggressive failover controla si las llamadas existentes se conmutan o solo las nuevas. Para informacion sobre pasarelas avanzadas, consulte nuestra guia de failover de pasarelas del sistema VOS3000. ๐ง
Diagnostico Paso a Paso ๐
Diagnosticar las VOS3000 llamadas cortadas requiere analizar los CDR y los codigos de finalizacion de las llamadas afectadas. Los codigos de finalizacion proporcionan informacion critica sobre por que se corto la llamada. Para informacion sobre codigos, consulte nuestra guia de codigos de finalizacion del sistema VOS3000. ๐
๐ Codigo Finalizacion
Significado
Causa Probable
๐ง Solucion
๐ Normal BYE
Una parte colgo
Fin normal de llamada
Verificar con usuario
๐ RTP Timeout
Sin flujo RTP
Problema de red/media
Ajustar RTP timeout
โฑ๏ธ Session Timeout
Sesion expirada
Session Timer no renovado
Configurar keepalive
๐ Switch/Failover
Cambio de ruta
Failover agresivo
Ajustar switch limit
๐ซ Proveedor rechaza
SIP 503/487
Proveedor sin capacidad
Failover a otro proveedor
๐ฅ Firewall
Sin BYE ni CANCEL
UDP timeout en firewall
Configurar NAT keepalive
Preguntas Frecuentes sobre VOS3000 Llamadas Cortadas โ
โ Por que se cortan las llamadas en VOS3000 despues de unos minutos?
Las VOS3000 llamadas cortadas despues de unos minutos son tipicamente causadas por firewall UDP timeout. Los firewalls eliminan las conexiones UDP inactivas despues de un periodo determinado (5-30 minutos). Si no hay trafico SIP de mantenimiento durante la llamada, la entrada se elimina y los paquetes posteriores se bloquean, cortando la llamada. La solucion es configurar SIP NAT keepalive en VOS3000 para enviar paquetes periodicos que mantengan la conexion activa en el firewall. ๐ฅ
โ Como evito que las llamadas se corten por RTP timeout?
Para evitar las VOS3000 llamadas cortadas por RTP timeout, ajuste el parametro RTP timeout en VOS3000 a un valor adecuado (30-60 segundos). Tambien verifique que el media proxy este habilitado para las pasarelas donde los dispositivos estan detras de NAT. Si el RTP timeout esta muy bajo, las pausas naturales en la conversacion pueden activar el timeout. Si esta muy alto, las llamadas fantasma consumiran recursos innecesariamente. โฑ๏ธ
โ El failover puede cortar llamadas existentes?
Si, el failover agresivo en VOS3000 puede cortar llamadas existentes cuando el sistema detecta degradacion en la ruta y conmuta a una ruta alternativa. Para evitar que el failover afecte llamadas en curso, configure el parametro aggressive failover para que solo afecte nuevas llamadas, no las existentes. Para informacion detallada sobre failover, consulte nuestra guia de failover del sistema VOS3000. ๐
โ Como verifico por que se corto una llamada en VOS3000?
Para determinar la causa de las VOS3000 llamadas cortadas, consulte los registros CDR en VOS3000. Los CDR contienen el codigo de finalizacion (release cause) que indica por que se corto la llamada. Busque el campo release cause en el CDR y comparelo con la tabla de codigos de finalizacion. Los codigos mas comunes para llamadas cortadas son RTP timeout, session timeout y SIP 503. Para informacion sobre CDR, consulte nuestra guia de registros CDR avanzados del sistema VOS3000. ๐
โ Que es el SIP NAT keepalive y como ayuda?
El SIP NAT keepalive es un mecanismo que envia paquetes SIP periodicos (tipicamente cada 20-30 segundos) para mantener las entradas NAT activas en los firewalls y routers. Sin keepalive, las VOS3000 llamadas cortadas ocurren porque el firewall elimina la entrada UDP asociada con la llamada. Para configurar NAT keepalive en VOS3000, acceda a los parametros SIP y establezca el intervalo de keepalive. Un intervalo de 20-30 segundos es adecuado para la mayoria de los firewalls. ๐
โ Las llamadas se cortan siempre a los 32 segundos, que significa?
Si las VOS3000 llamadas cortadas ocurren consistentemente a los 32 segundos, es casi seguro que se debe a un problema de negociacion de codec o a un firewall que bloquea el flujo RTP. El temporizador de 32 segundos es tipico del SIP INVITE timeout o del primer re-INVITE. Verifique que los codecs esten configurados correctamente en ambas pasarelas y que los puertos RTP esten abiertos en el firewall. ๐ต
Conclusion ๐
Las VOS3000 llamadas cortadas son un problema multifactorial que requiere un diagnostico sistematico basado en los codigos de finalizacion y el analisis de los flujos SIP y RTP. Con las configuraciones correctas de RTP timeout, Session Timer, NAT keepalive y failover, puede reducir drasticamente la tasa de llamadas cortadas y mejorar la experiencia del cliente. ๐
Para soporte profesional en la resolucion de problemas de llamadas cortadas, contactenos por WhatsApp al +8801911119966. Tambien puede descargar la ultima version desde vos3000.com/downloads. Para continuar aprendiendo, explore nuestros articulos sobre calidad QoS del sistema VOS3000 y registros CDR avanzados. ๐ค
Para consultas, contactenos por WhatsApp al +8801911119966. ๐ฑ
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๐ 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. โฑ๏ธ
๐ง 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.
๐ฏ 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
๐ What Is VOS3000 SIP Session Timer?
โฐ 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.
๐ก 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:
๐ Periodically sending re-INVITE or UPDATE messages to confirm the call is still alive
โ Automatically hanging up calls when no confirmation is received
๐ก๏ธ Preventing ultra-long bills caused by zombie sessions
๐ Detecting abnormal network conditions in real time
๐ RFC 4028 introduces the Session-Expires header and Min-SE header to SIP. Here’s how they map to VOS3000:
RFC 4028 Concept
VOS3000 Parameter
Function
Session-Expires
SS_SIP_SESSION_TTL
Total session lifetime before refresh required
Refresher negotiation
SS_SIP_SESSION_UPDATE_SEGMENT
Number of refresh attempts within TTL
Early termination
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP
Grace period before early hangup on no response
Non-timer fallback
SS_SIP_NO_TIMER_REINVITE_INTERVAL
Max call duration for non-session-timer UAs
โ๏ธ VOS3000 SIP Session Timer Parameters Deep Dive
๐ง Let’s examine each parameter in detail using the official VOS3000 2.1.9.07 manual data.
๐ SS_SIP_SESSION_TTL โ Detecting SIP Connected Status Interval
โฑ๏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.
Attribute
Value
๐ Parameter Name
SS_SIP_SESSION_TTL
๐ข Default Value
600 seconds (10 minutes)
๐ Unit
Seconds
๐ Description
If 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.
๐ก 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.
โ ๏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.
๐ 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.
Attribute
Value
๐ Parameter Name
SS_SIP_SESSION_UPDATE_SEGMENT
๐ข Default Value
2
๐ Range
2 โ 10
๐ Description
SIP Timer reinvite (update) Interval โ divides the TTL into segments
๐ฏ Calculation: The actual re-INVITE interval = SS_SIP_SESSION_TTL รท SS_SIP_SESSION_UPDATE_SEGMENT
TTL (seconds)
Segment
Re-INVITE Interval
Use Case
600
2
300s (5 min)
โ Default โ balanced
600
4
150s (2.5 min)
๐ง More frequent checks
600
6
100s (1.7 min)
๐ก Unstable networks
600
10
60s (1 min)
โ ๏ธ High overhead
1800
3
600s (10 min)
๐ Long calls, stable net
๐ก 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.
โฐ SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP โ Early Hangup Timer
๐ 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.
Attribute
Value
๐ Parameter Name
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP
๐ข Default Value
0 seconds (disabled)
๐ Unit
Seconds
๐ Description
SIP Timer no reinvite (update) Early Hang up โ extra grace period before terminating
โ ๏ธ When set to 0 (default): VOS3000 hangs up immediately when the session timer expires without confirmation. No grace period is given.
โ 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.
๐ก 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.
๐ฑ 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.
Attribute
Value
๐ Parameter Name
SS_SIP_NO_TIMER_REINVITE_INTERVAL
๐ข Default Value
7200 seconds (2 hours)
๐ Unit
Seconds
๐ Description
If SIP caller doesn’t support “timer”, softswitch will stop the call when the time is up
๐ 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.
โ ๏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).
๐ How VOS3000 SIP Session Timer Works โ Complete Flow
๐ Understanding the full session timer flow is essential for proper configuration. Here’s exactly what happens during a call:
๐ฏ Scenario A: Caller SUPPORTS Session Timer
๐ Call Established (200 OK)
โ
โโโ VOS3000 starts TTL countdown (SS_SIP_SESSION_TTL = 600s)
โ
โโโ At TTL/Segment = 300s โโโบ VOS3000 sends re-INVITE/UPDATE
โ โโโ โ 200 OK received โ Session confirmed, timer resets
โ โโโ โ No response โ Retry at next segment
โ
โโโ At TTL = 600s โโโบ Final check
โ โโโ โ 200 OK received โ Session confirmed, timer resets
โ โโโ โ No response โ Call terminated (BYE sent)
โ โโโ If EARLY_HANGUP > 0 โ Wait X seconds, then BYE
โ
โโโ ๐ Cycle repeats for duration of call
๐ฏ Scenario B: Caller Does NOT Support Session Timer
๐ Call Established (200 OK โ no Session-Expires header)
โ
โโโ VOS3000 detects no timer support
โ
โโโ No re-INVITE/UPDATE messages sent
โ
โโโ Call continues until...
โ โโโ ๐ฑ Normal BYE from either party, OR
โ โโโ โฐ Duration exceeds SS_SIP_NO_TIMER_REINVITE_INTERVAL (7200s)
โ โโโ VOS3000 forcibly terminates call (BYE sent)
โ
โโโ โ No active session detection possible
๐ Here’s the full reference table combining all session timer parameters from the official VOS3000 2.1.9.07 manual:
Parameter
Default
Unit
Range
Purpose
SS_SIP_SESSION_TTL
600
Seconds
โ
Session expiry detection interval
SS_SIP_SESSION_UPDATE_SEGMENT
2
Count
2โ10
Re-INVITE interval divider
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP
0
Seconds
โ
Grace period before early hangup
SS_SIP_NO_TIMER_REINVITE_INTERVAL
7200
Seconds
โ
Max call time for non-timer UAs
๐ก๏ธ Common VOS3000 SIP Session Timer Problems and Solutions
โ ๏ธ Even with proper configuration, session timer issues can arise. Here are the most common problems and their fixes:
โ Problem 1: Calls Dropping Every 5 Minutes
๐ Symptom: Active calls are being terminated at exactly the re-INVITE interval.
๐ก 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.
โ Solutions:
๐ง Increase SS_SIP_SESSION_TTL to give more time per cycle
๐ Reduce SS_SIP_SESSION_UPDATE_SEGMENT for fewer but longer intervals
๐ก Verify the far-end device supports RFC 4028 session timers
๐ Check if the far-end is behind a SIP ALG that drops re-INVITEs โ see our SIP debug guide
โ Problem 2: Ultra-Long Bills from Zombie Calls
๐ Symptom: CDR records show calls lasting hours beyond actual conversation time.
๐ก Cause: The SIP caller does not support session timers, and SS_SIP_NO_TIMER_REINVITE_INTERVAL is too high.
โ Solutions:
โฑ๏ธ Reduce SS_SIP_NO_TIMER_REINVITE_INTERVAL from 7200 to 1800 or lower
๐ Ensure SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP is set to 0 (immediate cleanup)
๐ Symptom: High CPU usage on VOS3000 server, excessive SIP signaling traffic.
๐ก Cause: SS_SIP_SESSION_UPDATE_SEGMENT is set too high, causing frequent re-INVITEs.
โ Solutions:
๐ Reduce SS_SIP_SESSION_UPDATE_SEGMENT to 2 (default) for fewer refresh attempts
โฑ๏ธ Increase SS_SIP_SESSION_TTL to 900 or 1800 for longer cycles
๐ง Balance detection speed against signaling load
๐ก VOS3000 SIP Session Timer Best Practices
๐ฏ Follow these best practices to get the most from your VOS3000 SIP session timer configuration:
Best Practice
Recommendation
Reason
๐ฏ Start with defaults
TTL=600, Segment=2
Proven balance for most deployments
๐ Monitor CDRs
Check for abnormally long calls weekly
Detects zombie calls early
๐ Lower non-timer limit
Set NO_TIMER to 1800โ3600
Reduces risk from non-RFC 4028 endpoints
๐ Test before production
Verify with SIP debug tools
Avoids unexpected call drops
๐ Verify endpoint support
Check Session-Expires in SIP INVITE
Confirms timer negotiation works
๐ก๏ธ Keep early hangup at 0
Unless network is very unstable
Immediate cleanup is safer
๐ก 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.
๐ VOS3000 SIP Session Timer and SIP Call Flow Interaction
๐ก The session timer operates within the broader SIP call flow. Understanding how it interacts with other SIP messages is critical:
๐ After configuration, verify that session timers are working correctly:
Using SIP Debug to Confirm Timer Negotiation ๐
# 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 <-- ๐ Session timer negotiated!
Min-SE: 90 <-- ๐ 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
๐ 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?
โฑ๏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?
๐ฑ 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 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?
๐ 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?
โ 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?
๐ก 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?
๐ก๏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.
๐ Need Expert Help with VOS3000 SIP Session Timer?
๐ง 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.
๐ฌ WhatsApp:+8801911119966 โ Get instant support for VOS3000 SIP session timer configuration, RFC 4028 compliance, and VoIP network optimization.
๐ 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. ๐
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VOS3000 Session Timer: Complete Guide to SIP Keep-Alive Configuration
VOS3000 session timer is a critical mechanism for maintaining call stability and preventing “zombie calls” that consume system resources. Based on RFC 4028 specifications, the session timer functionality in VOS3000 2.1.9.07 ensures that active VoIP sessions are properly monitored while failed or hung calls are detected and cleaned up automatically. This comprehensive guide covers all session timer parameters, NAT keep-alive configuration, and troubleshooting procedures based on the official VOS3000 manual.
๐ Need help configuring VOS3000 session timer? WhatsApp: +8801911119966
The VOS3000 session timer implements the SIP Session Timer mechanism defined in RFC 4028. This protocol extension addresses a fundamental problem in SIP-based VoIP systems: the inability to detect when a call has failed at one endpoint while the other endpoint believes the call is still active. These “zombie calls” can persist indefinitely, consuming system resources, occupying call capacity, and causing billing discrepancies.
VOS3000 provides a comprehensive set of session timer parameters that control how the softswitch monitors and maintains active SIP sessions. These parameters are configured in the System Parameters section and affect all SIP-based communications.
๐ Core Session Timer Parameters Table
โ๏ธ Parameter
๐ Default
๐ Range
๐ Description
๐ Manual Page
SS_SIP_SESSION_TTL
600
60-86400 sec
Detecting SIP connected status interval (Session-Expires value)
230
SS_SIP_SESSION_UPDATE_SEGMENT
2
2-10
Divisor for refresh interval calculation (TTL/segment)
NAT (Network Address Translation) devices maintain binding tables that map internal private IP addresses to external public addresses. These bindings have a timeout period, typically ranging from 30 to 300 seconds depending on the device. When a binding expires without traffic, incoming calls cannot reach the endpoint behind NAT.
๐ NAT Keep-Alive Parameters Table
โ๏ธ Parameter
๐ Default
๐ Range
๐ Function
๐ Page
SS_SIP_NAT_KEEP_ALIVE_MESSAGE
HELLO
Text string
Content of NAT keep-alive UDP packet
212
SS_SIP_NAT_KEEP_ALIVE_PERIOD
30
10-86400 sec
Interval between keep-alive transmissions
212
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL
500
1-10000 ms
Delay between individual keep-alive packets in batch
VOS3000 Debug Trace - Session Timer Analysis:
==============================================
Step 1: Enable Debug Trace
Navigation: System โ Debug trace
Enable: Check "On"
Set duration: 10-30 minutes
Step 2: Look for Session Timer Headers in SIP Messages:
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK12345
From: ;tag=abc123
To:
Call-ID: [email protected]
CSeq: 1 INVITE
Contact:
Session-Expires: 600;refresher=uac โ SESSION TIMER HEADER
Min-SE: 90 โ MINIMUM SESSION EXPIRES
Content-Type: application/sdp
Content-Length: ...
Step 3: Check 200 OK Response:
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
SIP/2.0 200 OK
...
Session-Expires: 600;refresher=uac โ CONFIRMED SESSION TIMER
...
Step 4: Look for Session Refresh Messages (UPDATE or re-INVITE):
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
UPDATE sip:[email protected]:5060 SIP/2.0
...
Session-Expires: 600 โ REFRESHING SESSION
...
Step 5: If No Session Timer Headers Found:
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
- Endpoint does not support RFC 4028
- VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL
- Maximum call duration will be enforced
๐ Session Timer vs NAT Keep-Alive Comparison
๐ Aspect
โฑ๏ธ Session Timer
๐ก NAT Keep-Alive
Primary Purpose
Detect failed calls, prevent zombie sessions
Maintain NAT bindings for incoming calls
RFC Standard
RFC 4028 (SIP Session Timer)
NAT traversal best practices
Protocol Used
SIP re-INVITE or UPDATE messages
UDP packets or SIP messages
When Active
During active call (after 200 OK)
While endpoint is registered
Direction
Bidirectional (negotiated refresh)
Server to endpoint (unidirectional)
Default Interval
600 seconds (10 minutes)
30 seconds
Failure Result
Call terminated, CDR updated
Incoming calls may fail
Endpoint Support Required
Yes (RFC 4028 compliance)
No (transparent to endpoint)
๐ฐ VOS3000 Installation and Support Services
Need professional help with VOS3000 session timer configuration? Our team provides comprehensive VOS3000 services including installation, configuration, and ongoing technical support.
โ Frequently Asked Questions about VOS3000 Session Timer
What happens if an endpoint doesn’t support session timer?
VOS3000 will use the SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter to limit the maximum call duration. This ensures that zombie calls cannot persist indefinitely even when the endpoint doesn’t support RFC 4028. Set this value based on your business requirements (default is 7200 seconds or 2 hours).
Why are my calls dropping exactly at 30 seconds?
30-second call drops are almost always caused by NAT binding timeout, not session timer issues. The solution is to enable NAT keep-alive by setting SS_SIP_NAT_KEEP_ALIVE_MESSAGE to a value like “HELLO” and reducing SS_SIP_NAT_KEEP_ALIVE_PERIOD to 15-20 seconds. Also check if SIP ALG is enabled on your router (it should be disabled).
What is the difference between re-INVITE and UPDATE for session refresh?
Both methods can be used for session refresh. UPDATE is generally preferred because it doesn’t modify the SDP session parameters, while re-INVITE also renegotiates media. VOS3000 automatically selects the appropriate method based on endpoint capabilities and configuration.
How do I calculate the optimal session timer refresh interval?
The refresh interval equals SS_SIP_SESSION_TTL divided by SS_SIP_SESSION_UPDATE_SEGMENT. With defaults (600 รท 2 = 300 seconds), VOS3000 sends a refresh every 5 minutes. For mobile networks, consider 300 รท 2 = 150 seconds for faster failure detection.
Can session timer prevent billing fraud?
Session timer helps prevent zombie calls that could result in incorrect CDR durations, but it’s not a fraud prevention mechanism. For fraud protection, implement proper account limits, IP restrictions, and monitor for unusual calling patterns using VOS3000’s built-in reports.
๐ Get Expert VOS3000 Session Timer Support
Need assistance configuring VOS3000 session timer or troubleshooting call drop issues? Our VOS3000 experts provide comprehensive support for session management, NAT traversal, and VoIP infrastructure optimization.
Understanding VOS3000 call termination reasons is essential for maintaining a reliable VoIP operation. When calls fail or disconnect unexpectedly, the termination reason in the CDR (Call Detail Record) provides crucial information for diagnosis. This comprehensive reference guide covers all server-side termination reasons, client-side error codes, and provides actionable troubleshooting steps based on the official VOS3000 2.1.9.07 manual documentation.
๐ Need help troubleshooting VOS3000 call failures? WhatsApp: +8801911119966
Table of Contents
๐ Understanding Call Termination in VOS3000
Every call processed through VOS3000 generates a CDR record that includes the termination reason. This information is captured at the “Termination Reason” or “Call End Reason” field and indicates why the call ended. Understanding these reasons helps identify patterns, troubleshoot recurring issues, and optimize call success rates.
๐ Where to Find Termination Information (VOS3000 Call Termination Reasons)
Navigation in VOS3000 Client:
1. Recent CDR
Location: Data Query โ Recent CDR
Purpose: View recent call records with termination info
2. CDR Query
Location: Data Query โ CDR
Purpose: Search historical CDR with filters
3. Call Analysis
Location: Operation Management โ Business Analysis โ Call Analysis
Purpose: Deep dive into specific call signaling
Key CDR Fields for Diagnosis:
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
โ Field โ Information Provided โ
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโค
โ Termination Reason โ Why the call ended (primary field) โ
โ Session Time โ How long call lasted โ
โ PDD โ Post dial delay โ
โ Caller/Callee IP โ Endpoint addresses โ
โ Codec โ Audio encoding used โ
โ Setup Time โ When call started โ
โ Connect Time โ When call was answered โ
โ End Time โ When call terminated โ
โโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโโ
Server-side termination reasons are generated by VOS3000 itself when the softswitch decides to end or reject a call. These reasons indicate specific conditions that prevented call completion. (VOS3000 Call Termination Reasons)
When calls involve H.323 protocol, termination reasons may include standard H.323 cause codes. These codes follow ITU-T Q.931 recommendations and provide detailed information about call failures.
๐ Common H.323 Cause Codes (VOS3000 Call Termination Reasons)
Cause Code
Name
Meaning
1
UnallocatedNumber
Number not assigned to any destination
3
NoRouteToDestination
No route to the called number
6
ChannelUnacceptable
Channel not acceptable for this call
16
NormalCallClearing
Call cleared normally
17
UserBusy
User is busy
18
NoResponse
No response from user
19
NoAnswer
User alerted but did not answer
21
CallRejected
Call was rejected
27
DestinationOutOfOrder
Destination cannot be reached
28
InvalidNumberFormat
Number format invalid
34
NoCircuitChannelAvailable
No channel available
38
NetworkOutOfOrder
Network not functioning properly
41
TemporaryFailure
Temporary network problem
42
Congestion
Network congestion
44
RequestedCircuitNotAvailable
Requested circuit not available
47
ResourceUnavailable
Insufficient resources
49
QoSNotAvailable
Requested QoS not available
๐ SIP Error Codes Reference
SIP responses follow standard HTTP-style status codes. Understanding these codes helps diagnose problems when they appear in CDR records or Call Analysis.
The most common reasons are “Hang-Up by Caller” and “Hang-Up by Called” which are normal terminations. For abnormal terminations, “Response Timeout” and “Connection Timeout” are most frequent, usually caused by network issues, firewall problems, or endpoint misconfiguration.
How do I differentiate between timeout types?
Response Timeout occurs when the called party doesn’t answer (no 180 Ringing or 200 OK). Connection Timeout occurs when SIP messages don’t receive any response after retries. Proceeding Timeout occurs during call setup when 100 Trying is received but no further progress. Session Timeout happens during an established call when session timer updates fail.
Why do I see “Insufficient Balance” for accounts with credit?
This can occur when: the account has credit but the rate for the destination is higher than the balance, there’s a minimum balance requirement configured, or the account’s overdraft limit has been reached. Check rate tables and account settings in Account Management.
What causes “No Matching Rate” errors?
This occurs when a call is made to a destination prefix that doesn’t have a corresponding entry in the rate table. Check that rate prefixes cover all destination patterns. Remember that VOS3000 uses longest prefix matching, so ensure appropriate prefix entries exist.
How do I troubleshoot intermittent “Session Timeout” errors?
Session Timeout typically indicates NAT binding expiry or SIP Timer issues. Check NAT keep-alive settings (SS_SIP_NAT_KEEP_ALIVE_PERIOD), verify session timer configuration (SS_SIP_SESSION_TTL), and ensure the client supports SIP Session Timers. If the client doesn’t support timers, check SS_SIP_NO_TIMER_REINVITE_INTERVAL for the maximum call duration.
๐ Get Expert Help with VOS3000 Troubleshooting
Need assistance diagnosing call failures or optimizing your VOS3000 performance? Our team provides comprehensive VOS3000 support, CDR analysis, and troubleshooting services.