VOS3000 Period Capacity Configuration, VOS3000 Period Dial Plan, VOS3000 RTP Interrupt Detection, VOS3000 Lowest Profit Rate Limit, VOS3000 Max Minute Rate Cap, VOS3000 Sort Lowest Rate Per Second, VOS3000 Check Rate Before Routing, VOS3000 Sort by Lowest Rate, VOS3000 Bilateral Reconciliation, VOS3000 SIP OPTIONS Online Check, VOS3000 T38 Fax Over IP, VOS3000 G729 Annex B Silence, VOS3000 Gateway Group Reserved Lines, VOS3000 Auxiliary Ring Tone

VOS3000 RTP Interrupt Detection Accurate Four-Mode Media Monitoring

VOS3000 RTP Interrupt Detection Accurate Four-Mode Media Monitoring

๐Ÿ“ก When a VoIP call is established and both parties are conversing, the media path is carrying RTP packets in both directions. But what happens when one direction stops โ€” the RTP stream is interrupted while the SIP signaling remains active? The caller hears silence, but the call never hangs up. The VOS3000 RTP interrupt detection feature solves this problem by monitoring RTP packet flow and taking action when media is lost, ensuring that silent zombie calls do not consume gateway resources or confuse billing records. ๐Ÿ”ง

โš™๏ธ According to the VOS3000 V2.1.9.07 Manual ยง2.5.1.1 (page 32), the VOS3000 RTP interrupt detection offers four modes: None (disable detection), Server to Remote (detect audio sent from server to device), Remote to Server (detect audio sent from device to server), and Bidirection (detect both sides โ€” if any one side has no audio, the call will be interrupted). Each mode provides a different level of monitoring granularity, allowing you to choose between passive observation and automatic call termination based on your deployment requirements. The VOS3000 RTP interrupt detection is a per-gateway setting configured in the Additional settings panel under Normal settings. ๐Ÿ“Š

๐ŸŽฏ This guide provides a complete, manual-verified reference for all four VOS3000 RTP interrupt detection modes. All parameter definitions are sourced exclusively from the official VOS3000 V2.1.9.07 Manual ยง2.5.1.1 (page 32). No fabricated values, no guesswork. For expert assistance, contact us on WhatsApp at +8801911119966. ๐Ÿ“˜

๐Ÿ” What Is VOS 3000 RTP Interrupt Detection?

๐Ÿ“‹ The VOS3000 RTP interrupt detection is a per-gateway feature that monitors RTP media packet flow during active calls and takes action when the RTP stream is interrupted. It is configured in the Routing Gateway Additional settings > Normal panel, in the “RTP interrupt detection” dropdown. The VOS3000 RTP interrupt detection requires the media proxy to be enabled for the call, as the softswitch can only observe RTP packets when it is proxying the media stream.

๐Ÿ’ก Key characteristics of RTP Interrupt Detection:

  • ๐Ÿ“ก Configuration location: Routing gateway > Additional settings > Normal > RTP interrupt detection
  • ๐Ÿ“‹ Per-gateway scope: Each routing gateway has its own VOS3000 RTP interrupt detection setting
  • ๐Ÿ”„ Four modes: None, Server to Remote, Remote to Server, Bidirection
  • ๐Ÿ”ง Prerequisite: Media proxy must be enabled (Auto, On, or Must On) for the VOS3000 RTP interrupt detection to function
  • ๐Ÿ“Š Action on detection: The call is interrupted (terminated) when RTP loss is detected in the monitored direction

๐Ÿ“‹ The Four VOS 3000 RTP Interrupt Detection Modes

๐Ÿ“Š The VOS3000 V2.1.9.07 Manual ยง2.5.1.1 (page 32) defines four modes for the VOS3000 RTP interrupt detection. Each mode determines which RTP direction is monitored and what action is taken when an interruption is detected:

ModeDescription (Manual)What Is MonitoredWhen Call Is Interrupted
๐Ÿšซ NoneDisable detectionNo RTP monitoringNever โ€” RTP interruptions are ignored
๐Ÿ“ค Server to RemoteDetect audio sent from server to deviceOutbound RTP: VOS3000 server โ†’ gateway deviceWhen server-to-device RTP stops flowing
๐Ÿ“ฅ Remote to ServerDetect audio sent from device to serverInbound RTP: gateway device โ†’ VOS3000 serverWhen device-to-server RTP stops flowing
๐Ÿ”„ BidirectionDetect both sides, if any one side no audio, the call will be interruptBoth directions simultaneouslyWhen RTP stops in EITHER direction

๐Ÿ’ก Critical manual note: The Bidirection mode description in the VOS3000 V2.1.9.07 Manual is explicit: “if any one side no audio, the call will be interrupt.” This means that losing RTP in just one direction is sufficient to trigger call interruption. This is the most aggressive VOS3000 RTP interrupt detection mode and provides the most comprehensive media monitoring, but it may also terminate calls prematurely in environments where temporary RTP gaps are normal (such as satellite links with variable latency).

๐Ÿ“Š Media Proxy Dependency for RTP Interrupt Detection

๐Ÿ”— The VOS3000 RTP interrupt detection requires the media proxy to be active for the call. Without media proxy, the softswitch does not observe the RTP stream and cannot detect interruptions. The VOS3000 RTP interrupt detection is fundamentally dependent on the media proxy configuration.

Media Proxy ModeRTP Interrupt Detection Effective?Implication for VOS3000 RTP Interrupt Detection
Auto (default)โœ… Yes โ€” when proxy is activatedVOS3000 RTP interrupt detection works for proxied calls; direct-RTP calls bypass monitoring
On / Must Onโœ… Yes โ€” all calls proxiedVOS3000 RTP interrupt detection monitors every call through this gateway
OffโŒ No โ€” no RTP observationVOS3000 RTP interrupt detection has no effect โ€” cannot observe RTP without proxy

๐Ÿ“Š Configuration recommendation: For the VOS3000 RTP interrupt detection to function reliably, set the media proxy to “Auto” or “On” for gateways where media monitoring is required. When media proxy is “Off,” the VOS3000 RTP interrupt detection parameter exists in the configuration but has no functional effect because the softswitch never sees the RTP packets. For comprehensive media proxy setup, see our VOS3000 RTP media guide.

๐Ÿ”„ Direction Monitoring Explained

๐Ÿ“ก Understanding the directional terminology in the VOS3000 RTP interrupt detection is essential for correct configuration. The terms “Server to Remote” and “Remote to Server” refer to the direction of RTP packet flow relative to the VOS3000 softswitch:

DirectionRTP FlowWhat Interruption IndicatesUse Case
๐Ÿ“ค Server to RemoteVOS3000 โ†’ Gateway device (outbound media)The originating side stopped sending audio โ€” caller may have gone silent or network issue on originating sideMonitor whether the calling party’s audio reaches the gateway
๐Ÿ“ฅ Remote to ServerGateway device โ†’ VOS3000 (inbound media)The gateway stopped sending audio โ€” gateway or downstream network issueMonitor whether the gateway’s audio reaches VOS3000
๐Ÿ”„ BidirectionBoth directionsEither side stopped โ€” comprehensive media loss detectionMaximum protection โ€” detects any RTP interruption regardless of direction

๐Ÿ’ก Direction selection tip: Choose “Server to Remote” when you want to detect if the calling party’s audio stops reaching the gateway. Choose “Remote to Server” when you want to detect if the gateway’s audio stops reaching VOS3000. Choose “Bidirection” for comprehensive VOS3000 RTP interrupt detection that catches media loss in either direction. For most production deployments, Bidirection provides the most complete monitoring, but it may be too aggressive for networks with occasional RTP gaps.

๐Ÿ“Š VOS 3000 RTP Interrupt Detection Mode Comparison by Deployment

DeploymentRecommended ModeRationale
๐Ÿข Retail VoIP (reliable networks)BidirectionClean up zombie calls quickly โ€” reliable network means RTP gaps indicate real problems
๐Ÿ”„ Wholesale terminationRemote to ServerFocus on whether the downstream carrier is delivering audio โ€” the most important quality indicator
๐Ÿ“ก Satellite / high-latency linksNone or Server to RemoteAvoid false positives from temporary RTP gaps on unreliable links
๐Ÿ’ณ Calling card / IVR servicesBidirectionZombie calls waste IVR resources and confuse billing โ€” aggressive cleanup is essential
๐Ÿงช Testing / lab environmentNoneDisable VOS3000 RTP interrupt detection during testing to avoid premature call termination

๐Ÿ›ก๏ธ Common VOSS3000 RTP Interrupt Detection Problems and Solutions

โŒ Problem 1: Calls Being Terminated Prematurely

๐Ÿ” Symptom: Active calls with normal conversation are being terminated unexpectedly, with CDR records indicating RTP interrupt detection as the cause.

๐Ÿ’ก Cause: The VOS3000 RTP interrupt detection is set to Bidirection mode on a network with occasional RTP packet gaps. Temporary network congestion or jitter buffers can cause brief RTP interruptions that trigger the detection and terminate the call.

โœ… Solutions:

  • ๐Ÿ”ง Change the VOS3000 RTP interrupt detection to a less aggressive mode โ€” Server to Remote or Remote to Server instead of Bidirection
  • ๐Ÿ“Š If both directions are needed, consider disabling VOS3000 RTP interrupt detection and using monitoring alarms instead for passive observation
  • ๐Ÿ“‹ Investigate network quality issues causing the RTP gaps โ€” resolve the root cause rather than adjusting the detection threshold

โŒ Problem 2: RTP Interrupt Detection Not Working

๐Ÿ” Symptom: The VOS3000 RTP interrupt detection is configured to Server to Remote or Bidirection, but calls with no audio continue without being terminated.

๐Ÿ’ก Cause: The media proxy is not enabled for the affected calls. Without media proxy, VOS3000 cannot observe RTP packets and the VOS3000 RTP interrupt detection has no effect.

โœ… Solutions:

  • ๐Ÿ”ง Verify the media proxy setting for the routing gateway โ€” must be Auto, On, or Must On
  • ๐Ÿ“Š Check if specific calls are bypassing the media proxy due to codec negotiation or NAT configuration
  • ๐Ÿ“‹ Ensure the RTP media proxy is functioning correctly for all calls through this gateway

โŒ Problem 3: One-Way Audio Not Detected

๐Ÿ” Symptom: One-way audio occurs on calls through a gateway with VOS3000 RTP interrupt detection enabled, but the call is not terminated โ€” the detection fails to catch the media loss.

๐Ÿ’ก Cause: The VOS3000 RTP interrupt detection mode is set to monitor only one direction (e.g., Server to Remote), but the audio loss is occurring in the opposite direction (Remote to Server). The unmonitored direction is not checked.

โœ… Solutions:

  • ๐Ÿ”ง Change the VOS3000 RTP interrupt detection to Bidirection mode to monitor both RTP directions
  • ๐Ÿ“Š Analyze CDR records to determine which direction typically loses audio โ€” adjust the VOS3000 RTP interrupt detection mode accordingly
  • ๐Ÿ“‹ Resolve the underlying one-way audio issue rather than relying solely on detection

๐Ÿ’ก VOS 3000 RTP Interrupt Detection Best Practices

Best PracticeRecommendationReason
๐Ÿ“ก Enable media proxy firstSet media proxy to Auto or On before configuring the VOS3000 RTP interrupt detection๐Ÿ”ง VOS3000 RTP interrupt detection cannot function without media proxy observing RTP packets
๐Ÿ”„ Use Bidirection for clean networksEnable Bidirection mode when network quality is reliable๐Ÿ“Š Most comprehensive VOS3000 RTP interrupt detection โ€” catches media loss in either direction
โš ๏ธ Be cautious on unstable linksUse None or single-direction detection on satellite or high-jitter links๐Ÿ“‹ Prevents false-positive call terminations from temporary RTP gaps
๐Ÿ“Š Monitor CDR after enablingCheck call end reasons in CDR after deploying the VOS3000 RTP interrupt detection๐Ÿ“ˆ Verifies detection is working correctly and not causing premature terminations
๐Ÿ“ž Pair with RTP lock-inKeep SS_GATEWAY_SWITCH_STOP_AFTER_RTP_START = On alongside the VOS3000 RTP interrupt detection๐Ÿ›ก๏ธ RTP lock-in prevents switching after media starts; VOS3000 RTP interrupt detection monitors for media loss

๐Ÿ’ฌ Need VOS3000 RTP detection help? WhatsApp +8801911119966

๐Ÿ“‹ VOS3000 RTP Interrupt Detection Quick Decision Table

๐ŸŽฏ Use this decision table to select the correct VOS3000 RTP interrupt detection mode for your deployment scenario:

Your RequirementRecommended ModeReason
I want to monitor for audio loss without disconnecting callsServer to remote or Remote to serverDetects one-way audio and reports without terminating the call
I need to automatically hang up calls with no audioBidirectionDetects loss in both directions and terminates dead calls automatically
My gateway reports false RTP interrupts frequentlyNoneDisables detection to prevent false positives from unreliable gateways

โ“ Frequently Asked Questions

โ“ What is the default VOS 3000 RTP interrupt detection setting?

๐Ÿ”ง The default VOS3000 RTP interrupt detection setting depends on the gateway configuration. For new routing gateways, the default is typically “None” (detection disabled), meaning the softswitch does not monitor RTP packet flow for interruptions. You must explicitly configure the VOS3000 RTP interrupt detection to one of the active modes (Server to Remote, Remote to Server, or Bidirection) to enable media monitoring for that gateway. The VOS3000 RTP interrupt detection is a per-gateway setting, so you can enable different modes on different gateways based on their network characteristics.

โ“ Does VOS3000 RTP interrupt detection work without media proxy?

๐Ÿ“ก No, the VOS3000 RTP interrupt detection requires the media proxy to be active for the call. Without media proxy, VOS3000 does not see the RTP packets flowing between the caller and the gateway, so it cannot detect when the RTP stream is interrupted. The VOS3000 RTP interrupt detection is fundamentally dependent on the media proxy’s ability to observe and relay RTP media packets. If the media proxy is set to “Off,” the VOS3000 RTP interrupt detection parameter exists in the gateway configuration but has no functional effect.

โ“ What is the difference between Bidirection and single-direction detection?

๐Ÿ”„ The key difference is the scope of monitoring. “Server to Remote” only monitors the outbound RTP direction (VOS3000 โ†’ gateway device). “Remote to Server” only monitors the inbound RTP direction (gateway device โ†’ VOS3000). “Bidirection” monitors both directions simultaneously โ€” according to the manual, “if any one side no audio, the call will be interrupt.” The VOS3000 RTP interrupt detection in Bidirection mode provides the most comprehensive monitoring but is also the most aggressive, as it will terminate the call if RTP stops in either direction. Single-direction detection is more tolerant of one-way RTP gaps.

โ“ Will RTP interrupt detection terminate calls during temporary network congestion?

โš ๏ธ Yes, the VOS3000 RTP interrupt detection can terminate calls during temporary network congestion if the RTP gap exceeds the detection threshold. This is why the VOS3000 RTP interrupt detection should be used cautiously on networks with variable quality. For reliable, low-latency networks, Bidirection mode works well. For networks prone to temporary congestion or jitter, consider using “Server to Remote” or “None” mode, and instead rely on ASR ACD analysis and quality monitoring to detect media problems without automatically terminating calls.

โ“ Can I set different RTP interrupt detection modes on different gateways?

๐Ÿ”ง Yes, the VOS3000 RTP interrupt detection is configured per-gateway. Each routing gateway can have its own VOS3000 RTP interrupt detection mode. This means you can set Bidirection mode on a gateway connected to a reliable carrier, Server to Remote on a gateway with occasional upstream issues, and None on a gateway connected through a satellite link. This per-gateway flexibility in the VOS3000 RTP interrupt detection system allows you to tailor media monitoring to the specific network conditions of each gateway.

โ“ How does RTP interrupt detection interact with gateway failover?

๐Ÿ”„ The VOS3000 RTP interrupt detection and gateway failover operate at different levels. The VOS3000 RTP interrupt detection monitors media flow on an established call and terminates it if RTP is lost. Gateway failover (controlled by SS_GATEWAY_SWITCH_LIMIT and related parameters) handles the pre-connect phase where VOS3000 tries alternative gateways when the initial attempt fails. Once a call is connected and RTP is flowing, the VOS3000 RTP interrupt detection takes over to monitor media health. If the VOS3000 RTP interrupt detection terminates the call due to RTP loss, no further failover occurs โ€” the call is ended, not retried. For failover configuration, see our vendor failover setup guide. Need help? Contact us on WhatsApp at +8801911119966. ๐Ÿ“ฑ

๐Ÿ“ž Need Expert Help with VOS 3000 RTP Interrupt Detection?

๐Ÿ”ง Proper VOS3000 RTP interrupt detection configuration is essential for preventing zombie calls, conserving gateway resources, and maintaining accurate billing records. The VOS3000 RTP interrupt detection system with its four modes provides the flexibility to match monitoring intensity to network reliability. Whether you are troubleshooting one-way audio issues, configuring media monitoring for the first time, or balancing detection sensitivity against false positives on unreliable links, expert guidance ensures your VOS3000 RTP interrupt detection delivers the right level of protection. ๐Ÿ“ก

๐Ÿ’ฌ WhatsApp: +8801911119966 โ€” Get immediate assistance with VOS3000 RTP interrupt detection configuration, VOS3000 RTP interrupt detection troubleshooting, media proxy setup, and one-way audio resolution. Our team specializes in VOS3000 call quality management, media monitoring, and carrier-grade VoIP deployment. ๐Ÿ”ง

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


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Detecciรณn interrupciรณn RTP VOS3000 Accurate monitoreo de medios en cuatro modos

Detecciรณn interrupciรณn RTP VOS3000 Accurate monitoreo de medios en cuatro modos

La detecciรณn interrupciรณn RTP VOS3000 es uno de los mecanismos mรกs crรญticos del mapping gateway para garantizar la calidad y la eficiencia de las llamadas VoIP en producciรณn. Cuando el flujo RTP (Real-Time Transport Protocol) que transporta el audio de una llamada se interrumpe inesperadamente โ€” por fallas de red, desconexiรณn del gateway o problemas de NAT โ€” la llamada puede quedar en estado zombi sin audio para ninguna de las partes. VOS3000 ofrece cuatro modos de detecciรณn que permiten desde ignorar el problema hasta colgar automรกticamente la llamada afectada, liberando recursos y evitando CDRs con duraciones infladas. ยฟNecesita ayuda con esta configuraciรณn? Contรกctenos por WhatsApp: +8801911119966.

En redes VoIP de alta densidad, especialmente en entornos de wholesale donde miles de llamadas concurrentes atraviesan mรบltiples gateways y operadores, la interrupciรณn del flujo RTP es un evento frecuente. Las llamadas sin audio consumen canales concurrentes, generan CDRs con duraciones que no reflejan comunicaciรณn real y degradan la experiencia del usuario final. Segรบn el manual de administraciรณn del mapping gateway (ยง2.5.1.1, pรกg. 87-89, apartado Media Parameters), VOS3000 proporciona un mecanismo robusto para detectar y responder a estas interrupciones de manera automรกtica segรบn la estrategia que el operador defina para cada gateway de forma independiente.

๐Ÿ“‹ Los Cuatro Modos de Detecciรณn Interrupciรณn RTP

VOS3000 proporciona cuatro modos distintos de detecciรณn, cada uno diseรฑado para un escenario operativo diferente. La selecciรณn del modo se configura de forma independiente para cada mapping gateway, lo que permite aplicar estrategias diferentes segรบn el comportamiento de cada carrier.

๐Ÿ”น Modo๐Ÿ”น Nombre๐Ÿ”น Comportamiento๐Ÿ”น Caso de Uso
0None (Ninguno)No detecta interrupciones RTPEntornos de prueba o trรกfico interno confiable
1Detect Only (Solo Detectar)Detecta pero no realiza ninguna acciรณnMonitoreo estadรญstico interno
2Detect and Report (Detectar e Informar)Detecta y genera alarma visible en NOCMonitoreo activo con notificaciรณn
3Detect and Hang Up (Detectar y Colgar)Detecta y termina la llamada automรกticamenteRecuperaciรณn automรกtica de recursos

๐Ÿ” Modo None โ€” Sin Detecciรณn

Cuando el parรกmetro estรก configurado en modo None, el softswitch ignora completamente las interrupciones del flujo RTP. Si el audio deja de fluir, la llamada permanece activa hasta que una de las partes cuelga o se agota el session timer. Este modo es apropiado para entornos donde las interrupciones son extremadamente raras, como redes internas cerradas con gateways de alta confiabilidad. Detecciรณn Interrupciรณn RTP

โš ๏ธ Advertencia: En modo None, las llamadas sin audio pueden permanecer activas indefinidamente si el session timer no estรก configurado. Esto consume canales concurrentes innecesariamente y genera CDRs con duraciones infladas. Para la mayorรญa de entornos de producciรณn se recomienda al menos el modo Detect and Report. Detecciรณn Interrupciรณn RTP

๐Ÿ”น Aspecto๐Ÿ”น Comportamiento en Modo None
Detecciรณn RTPCompletamente deshabilitada
Llamada sin audioPermanece activa hasta colgar manual o timeout
Impacto en CDRDuraciรณn inflada โ€” se factura tiempo sin comunicaciรณn
Consumo de recursosCanal concurrente ocupado innecesariamente

๐Ÿ“Š Modo Detect Only โ€” Detecciรณn Sin Acciรณn

En el modo Detect Only, VOS3000 monitorea activamente el flujo RTP y detecta cuando se interrumpe, pero no realiza ninguna acciรณn automรกtica. La detecciรณn se registra internamente para fines estadรญsticos y puede utilizarse con las herramientas de monitoreo del sistema para generar reportes de calidad. Este modo es รบtil cuando el operador desea recopilar datos sobre la frecuencia de interrupciones sin afectar las llamadas. Para configuraciรณn de monitoreo avanzado, consulte nuestra guรญa de monitoreo y alarmas.

๐Ÿ”” Modo Detect and Report โ€” Detecciรณn con Notificaciรณn

El modo Detect and Report genera una alarma visible en el sistema de monitoreo del VOS3000 client cuando se detecta una interrupciรณn RTP. Esto permite a los operadores del NOC recibir alertas en tiempo real sobre llamadas afectadas, facilitando la identificaciรณn proactiva de problemas de red. La llamada permanece activa a pesar de la alarma, por lo que las partes pueden reanudar la comunicaciรณn si la interrupciรณn es temporal. Para entender mejor el sistema de alarmas, consulte nuestra guรญa de monitoreo. ยฟNecesita asesorรญa? Escrรญbanos por WhatsApp: +8801911119966.

๐Ÿ”น Modo๐Ÿ”น Detecciรณn๐Ÿ”น Acciรณn Automรกtica๐Ÿ”น Notificaciรณn๐Ÿ”น CDR
NoneNoNingunaNoDuraciรณn inflada
Detect OnlySรญNingunaSolo registro internoDuraciรณn inflada
Detect and ReportSรญNingunaAlarma visible en NOCDuraciรณn inflada
Detect and Hang UpSรญTermina la llamadaCDR registradoDuraciรณn precisa

๐Ÿ“ž Modo Detect and Hang Up โ€” Recuperaciรณn Automรกtica

El modo Detect and Hang Up es la opciรณn mรกs robusta y la mรกs recomendada para entornos de producciรณn de alta densidad. Cuando se detecta una interrupciรณn RTP, VOS3000 termina automรกticamente la llamada enviando un mensaje BYE a ambas partes, liberando canales concurrentes y recursos de medios, y generando un CDR con la duraciรณn real de la comunicaciรณn. Para entender cรณmo estos eventos se registran en los CDRs, consulte nuestra guรญa de anรกlisis de CDR. Para problemas de llamadas sin audio, vea tambiรฉn VOS3000 No Media Hangup.

โš™๏ธ Configuraciรณn Paso a Paso – Detecciรณn Interrupciรณn RTP

Para configurar la detecciรณn interrupciรณn RTP VOS3000 en un mapping gateway, siga estos pasos. La configuraciรณn se realiza de forma independiente para cada gateway. Para mรกs detalles sobre la configuraciรณn de gateways, consulte nuestra guรญa de configuraciรณn de gateways.

๐Ÿ”น Paso๐Ÿ”น Acciรณn๐Ÿ”น Detalle
1Abrir mapping gatewayGateway > Mapping Gateway en el cliente VOS3000
2Localizar RTP Interrupt DetectionSecciรณn Media Parameters, ยง2.5.1.1, pรกg. 87
3Seleccionar modoNone, Detect Only, Detect and Report, o Detect and Hang Up
4Configurar timeout RTPSegundos sin RTP antes de considerar interrupciรณn
5Guardar configuraciรณnAplicar cambios al gateway
6Verificar con pruebasSimular interrupciรณn RTP y verificar comportamiento

๐Ÿ“ˆ Impacto en la Facturaciรณn y Precisiรณn de CDRs

La elecciรณn del modo de detecciรณn tiene un impacto directo en la precisiรณn de los registros de facturaciรณn. Cuando una llamada pierde audio pero permanece activa (modos None, Detect Only o Detect and Report), el CDR continรบa acumulando duraciรณn hasta que la llamada se termina por otro medio. Esto resulta en CDRs con duraciones mayores al tiempo real de comunicaciรณn. El modo Detect and Hang Up resuelve este problema terminando la llamada en el momento de la detecciรณn. Para mรกs informaciรณn, consulte nuestra guรญa del sistema de facturaciรณn.

๐Ÿ”น Escenario๐Ÿ”น Sin Detecciรณn (None)๐Ÿ”น Con Detect and Hang Up
Llamada de 3 min con RTP interrumpido a los 30 segCDR registra 3 minutosCDR registra 30 segundos
Sobrecoste al cliente5x mรกs de lo realFacturaciรณn precisa
Disputa de facturaciรณnProbableMinimizada

๐Ÿ› ๏ธ Soluciรณn de Problemas Comunes – Detecciรณn Interrupciรณn RTP

Los siguientes problemas y soluciones le ayudarรกn a optimizar la configuraciรณn. Para problemas relacionados con audio, consulte nuestra guรญa de soluciรณn de eco y retardo.

๐Ÿ”น Problema๐Ÿ”น Causa Probable๐Ÿ”น Soluciรณn
Llamadas se cortan prematuramenteTimeout RTP demasiado cortoAumentar timeout a 30-60 seg
Llamadas sin audio permanecen activasModo configurado como NoneCambiar a Detect and Hang Up
Falsas detecciones en redes con alta latenciaLatencia variable excede el timeoutAjustar timeout segรบn latencia de red
No se detectan interrupcionesMedia proxy puede enmascarar interrupcionesVerificar configuraciรณn de media proxy
Cortes con codecs con supresiรณn de silencioVAD (G.729 Annex B) no envรญa paquetes en silencioDeshabilitar VAD o aumentar timeout

๐Ÿ”— Recursos Relacionados – Detecciรณn Interrupciรณn RTP en VOS3000

โ“ Preguntas Frecuentes sobre la Detecciรณn Interrupciรณn RTP en VOS3000

ยฟQuรฉ es la detecciรณn interrupciรณn RTP en VOS3000?

Es una funciรณn del mapping gateway de VOS3000 que monitorea el flujo de paquetes RTP durante una llamada activa. Cuando se detecta que los paquetes RTP han dejado de llegar durante un perรญodo configurable, el sistema puede tomar acciones que van desde registrar el evento hasta terminar automรกticamente la llamada. Esta funciรณn estรก documentada en la secciรณn ยง2.5.1.1 (pรกg. 87-89) del manual de administraciรณn y se configura de forma independiente para cada mapping gateway, permitiendo estrategias diferentes segรบn el comportamiento de cada carrier conectado a la plataforma.

ยฟCuรกl es la diferencia entre Detect and Report y Detect and Hang Up?

En modo Detect and Report, VOS3000 detecta la interrupciรณn y genera una alarma visible en el sistema de monitoreo del NOC, pero la llamada permanece activa โ€” las partes pueden seguir sin audio hasta que cuelguen manualmente. En modo Detect and Hang Up, VOS3000 detecta la interrupciรณn y termina inmediatamente la llamada enviando BYE a ambas partes, liberando recursos y generando un CDR con la duraciรณn real. El modo Detect and Hang Up es el recomendado para producciรณn wholesale porque garantiza CDRs precisos y evita el consumo innecesario de canales concurrentes. Detecciรณn Interrupciรณn RTP

ยฟQuรฉ valor de timeout RTP debo configurar?

Un valor tรญpico para entornos de producciรณn es de 30 a 60 segundos โ€” permite tolerar pausas breves en el audio sin generar falsas detecciones, pero es suficientemente corto para identificar interrupciones reales. En redes con alta latencia o enlaces satelitales, puede ser necesario aumentar el timeout a 90 o 120 segundos. El valor รณptimo debe determinarse mediante pruebas en su entorno especรญfico, ajustando segรบn la frecuencia de falsos positivos y la velocidad de detecciรณn deseada.

ยฟPuede la detecciรณn causar cortes de llamadas legรญtimas?

Sรญ, si el timeout RTP estรก configurado con un valor demasiado bajo, las pausas naturales en la conversaciรณn pueden ser detectadas errรณneamente como interrupciones RTP. Esto es mรกs probable cuando se utilizan codecs con supresiรณn de silencio (VAD) como G.729 Annex B, donde los perรญodos de silencio no generan paquetes RTP. Para evitar este problema, asegรบrese de que el timeout sea suficientemente largo (mรญnimo 30 segundos) y considere deshabilitar la supresiรณn de silencio en los gateways cuando utilice el modo Detect and Hang Up.

ยฟCรณmo afecta el media proxy a la detecciรณn de interrupciones RTP?

Cuando el media proxy estรก habilitado, el flujo RTP pasa a travรฉs del servidor VOS3000 en lugar de fluir directamente entre los endpoints. Esto puede facilitar la detecciรณn de interrupciones porque VOS3000 ve los paquetes de ambos lados. Sin embargo, si el media proxy estรก en modo bypass, los paquetes RTP no pasan por el servidor y la detecciรณn puede no funcionar correctamente. Siempre verifique la configuraciรณn del media proxy al implementar esta funciรณn.

ยฟEs recomendable usar el mismo modo para todos los gateways?

No necesariamente. La detecciรณn se configura por mapping gateway, lo que permite usar modos diferentes segรบn el comportamiento de cada carrier. Un gateway con un carrier de alta calidad puede usar Detect and Report para monitoreo pasivo, mientras que un gateway con un carrier propenso a interrupciones puede usar Detect and Hang Up para recuperaciรณn automรกtica. Esta flexibilidad permite optimizar la estrategia de monitoreo para cada interconexiรณn de forma independiente.

๐Ÿš€ Soporte Profesional

La configuraciรณn correcta de la detecciรณn interrupciรณn RTP es fundamental para mantener la calidad del servicio, la precisiรณn de la facturaciรณn y la eficiencia del uso de recursos en su plataforma VoIP. Nuestro equipo de especialistas puede ayudarle a seleccionar e implementar el modo รณptimo para cada gateway. Contรกctenos por WhatsApp: +8801911119966.

Desde la configuraciรณn inicial hasta la optimizaciรณn avanzada de calidad de servicio y monitoreo de alarmas, proporcionamos soporte experto para operadores en Colombia, Perรบ y toda Latinoamรฉrica. Escrรญbanos hoy al +8801911119966 y proteja la calidad de cada llamada en su red.


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

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

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

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

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

What Are Ghost Calls in VoIP?

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

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

Why Ghost Calls Are a Serious Problem

Ghost calls create multiple layers of problems for VoIP operators:

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

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

How Ghost Calls Occur: Causes and Symptoms

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

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

How VOS3000 No Media Hangup Works

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

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

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

The Auto-Disconnect Process Step by Step

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

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

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

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

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

Configuring SS_NOMEDIAHANGUPTIME in VOS3000

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

To configure SS_NOMEDIAHANGUPTIME, follow these steps:

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

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

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

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

Setting the Appropriate Timeout

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

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

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

No Media Hangup vs Session Timer: Critical Differences

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

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

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

Media Proxy Mode Interaction with No Media Hangup

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

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

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

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

Detecting Ghost Calls in CDR: Identifying the Patterns

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

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

Using Current Call Monitor for Real-Time Detection

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

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

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

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

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

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

Use Case: Preventing Billing Disputes from Ghost Calls

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

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

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

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

Use Case: Freeing Up Concurrent Call Capacity During Network Issues

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

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

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

Troubleshooting: Legitimate Calls Being Disconnected

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

Symptoms of Incorrect Disconnection

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

Resolving Incorrect Disconnection

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

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

Configuration and Testing Checklist (VOS3000 no media hangup)

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

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

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

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

FAQ: VOS3000 No Media Hangup

1. What is no media hangup in VOS3000?

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

2. What is the SS_NOMEDIAHANGUPTIME parameter?

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

3. How do ghost calls affect VoIP billing?

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Conclusion – VOS3000 no media hangup

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

Key takeaways from this guide:

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

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


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