⏱️ When VOS3000 sends a SIP INVITE, it enters a carefully timed sequence of timeout stages — each governed by a specific parameter that controls how long the softswitch waits at that phase before moving on or giving up. Understanding the complete VOS3000 SIP call progress timeout chain is essential for any VoIP operator who wants to eliminate mysterious call failures, optimize gateway channel utilization, and deliver a reliable calling experience. 📞
🔄 The call progress timeout chain consists of four critical parameters that fire sequentially during SIP call setup: SS_SIP_TIMEOUT_TRYING (20 seconds), SS_SIP_TIMEOUT_SESSION_PROGRESS (20 seconds), SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120 seconds), and SS_SIP_TIMEOUT_RINGING (120 seconds). Together with the initial SS_SIP_TIMEOUT_INVITE (10 seconds) timer, these five parameters define the entire timeout behavior from INVITE to answer. 🎯
🔧 This guide covers every parameter in the VOS3000 SIP call progress timeout chain — from the first 100 Trying response through Session Progress and Ringing stages to final answer or timeout failure. We explain how each timer works, when it fires, how per-gateway overrides give you granular control, and how to troubleshoot the most common timeout-related issues. All data is sourced exclusively from the official VOS3000 V2.1.9.07 Manual, Section 4.3.5.2 (Tables 4-3 and 4-4) — no guesses, no fabricated values. For expert assistance, contact us on WhatsApp at +8801911119966. 💡
📡 The VOS3000 SIP call progress timeout refers to the complete family of SIP timers that govern how long VOS3000 waits at each stage of the call setup process after sending an INVITE. These timers monitor provisional (1xx) SIP responses — the intermediate signals that indicate the call is progressing toward an answer. When a timer expires without the expected progress, VOS3000 terminates the call attempt and records the failure in the CDR. ⏱️
⚠️ Misconfiguring any of these timers can cause a range of problems: calls that disappear silently after 100 Trying, early media sessions that get cut off at 20 seconds, endless ringing that wastes gateway channels, and no-answer call forwarding that never triggers. Understanding how the complete chain works together is the key to avoiding these issues. 📋
📊 The VOS3000 SIP call progress timeout operates within a sequential chain. Each timer monitors a specific stage and hands off to the next when the call progresses. Here is the complete flow: 📡
📞 VOS3000 SIP Call Setup Timeout Chain — Complete Flow:
VOS3000 ──── INVITE ────► Destination
│
├── ⏱️ Timer 1: SS_SIP_TIMEOUT_INVITE (10s)
│ └── Waiting for ANY response to INVITE
│ ├── ❌ No response in 10s → Call failed (INVITE timeout)
│ └── ✅ 100 Trying received → Timer 1 stops, Timer 2 starts
│
├── ⏱️ Timer 2: SS_SIP_TIMEOUT_TRYING (20s) ◄── CALL PROGRESS
│ └── Waiting for progress beyond 100 Trying
│ ├── ❌ No 180/183/200 in 20s → Call failed (trying timeout)
│ └── ✅ 183 Session Progress received → Timer 2 stops
│ ├── 183 WITHOUT SDP → Timer 3a starts
│ └── 183 WITH SDP → Timer 3b starts
│
├── ⏱️ Timer 3a: SS_SIP_TIMEOUT_SESSION_PROGRESS (20s) ◄── CALL PROGRESS
│ └── 183 without SDP — no media path established
│ ├── ❌ No 180/200 in 20s → Call failed (session progress timeout)
│ └── ✅ 180 Ringing or 200 OK → Timer stops
│
├── ⏱️ Timer 3b: SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s) ◄── CALL PROGRESS
│ └── 183 with SDP — early media active (caller hears audio)
│ ├── ❌ No 180/200 in 120s → Call failed (early media timeout)
│ └── ✅ 180 Ringing or 200 OK → Timer stops
│
├── ⏱️ Timer 4: SS_SIP_TIMEOUT_RINGING (120s) ◄── CALL PROGRESS
│ └── 180 Ringing received — waiting for answer
│ ├── ❌ No 200 OK in 120s → CANCEL, no-answer
│ └── ✅ 200 OK → Call established! 🎉
│
└── 🔁 Post-answer: SIP Session Timer takes over
🔑 Key insight: Timers 2, 3a, 3b, and 4 are the VOS3000 SIP call progress timeout parameters. They only activate after VOS3000 receives at least one provisional response. If the gateway never responds at all, only Timer 1 (SS_SIP_TIMEOUT_INVITE) applies. For a complete breakdown of all SIP message flows, refer to our SIP call flow guide. 📡
📊 Here is the master reference table for all four VOS3000 SIP call progress timeout parameters, sourced from the official VOS3000 2.1.9.07 manual: 🔗
| Parameter | Default | Unit | Triggered By | Per-GW Override |
|---|---|---|---|---|
| SS_SIP_TIMEOUT_TRYING | 20 | Seconds | 100 Trying received, no further progress | Yes — Trying timeout |
| SS_SIP_TIMEOUT_SESSION_PROGRESS | 20 | Seconds | 183 without SDP received | Yes — SessionProgress(183) timeout |
| SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP | 120 | Seconds | 183 with SDP (early media) received | Yes — SessionProgress(SDP) timeout |
| SS_SIP_TIMEOUT_RINGING | 120 | Seconds | 180 Ringing received | Yes — Ringing timeout field |
📍 All SIP parameters are located at: Navigation → Operation management → Softswitch management → Additional settings → SIP parameter
⚡ Why do SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP and SS_SIP_TIMEOUT_RINGING have 120-second defaults while the other two are only 20 seconds? The answer is early media and active call progress. When a 183 response includes SDP or a 180 Ringing is received, audio is flowing — the caller is actively engaged with ringback, IVR announcements, or queue music. VOS3000 gives these calls 120 seconds because real audio is being exchanged. By contrast, a 100 Trying or 183 without SDP means no media is flowing — just a stalled signaling state that should time out quickly. 🎵
📞 The SS_SIP_TIMEOUT_TRYING parameter defines the maximum number of seconds VOS3000 will wait for call progress after receiving a 100 Trying provisional response. When VOS3000 sends a SIP INVITE and the far end replies with 100 Trying (meaning “I received your request and am processing it”), the trying timer starts. If no further progress signal arrives within the configured timeout — no 180 Ringing, no 183 Session Progress, no 200 OK — VOS3000 terminates the call attempt. ⏱️
| Attribute | Value |
|---|---|
| 📌 Parameter Name | SS_SIP_TIMEOUT_TRYING |
| 🔢 Default Value | 20 |
| 📐 Unit | Seconds |
| 📝 Description | SIP Trying timeout. Default value in “Routing Gateway > Additional settings > Protocol > SIP” |
| 📍 Navigation | Operation management → Softswitch management → Additional settings → SIP parameter |
💡 Key insight: The 100 Trying response is informational — it tells VOS3000 that the INVITE was received, but it does not indicate that the call is progressing. The trying timeout ensures that VOS3000 does not wait indefinitely for a dead-end gateway that acknowledged the INVITE but cannot process it further. This is a hop-by-hop response — it is not forwarded beyond the immediate SIP hop, which means the 100 Trying VOS3000 receives is from the next-hop gateway, not necessarily the ultimate destination.
📡 The SS_SIP_TIMEOUT_SESSION_PROGRESS parameter controls how long VOS3000 waits after receiving a 183 Session Progress response that does not contain an SDP body. A 183 without SDP indicates that the far end is processing the call but has not yet established a media path. 🔧
| Attribute | Value |
|---|---|
| 📌 Parameter Name | SS_SIP_TIMEOUT_SESSION_PROGRESS |
| 🔢 Default Value | 20 |
| 📐 Unit | Seconds |
| 📝 Description | SIP Session Progress (183) timeout. Default value in “Routing Gateway > Additional settings > Protocol > SIP” |
🔍 When does this timer apply? Some SIP servers and gateways send a 183 Session Progress without SDP as an intermediate response — for example, when the call is being routed through multiple hops or when the destination is being located. Since no media is established, this state should not persist long. The default of 20 seconds ensures VOS3000 moves on quickly if the call cannot progress. Unlike 100 Trying, the 183 is an end-to-end response — it comes from further downstream in the call path. ⏱️
🔊 The SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP parameter controls how long VOS3000 waits after receiving a 183 Session Progress with SDP. This is fundamentally different from the other two progress timeouts because SDP means a media path has been negotiated — audio is flowing even though the call is not yet answered. 🎶
| Attribute | Value |
|---|---|
| 📌 Parameter Name | SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP |
| 🔢 Default Value | 120 |
| 📐 Unit | Seconds |
| 📝 Description | SIP Session Progress with SDP timeout. Default value in “Routing Gateway > Additional settings > Protocol > SIP” |
📞 Common early media scenarios:
💡 Why 120 seconds? Early media calls are active audio sessions — the caller is hearing something, which means they are engaged. Cutting these off too early would terminate calls where the caller is listening to an IVR menu or waiting in a queue. The 120-second default provides ample time for these scenarios while still preventing runaway calls. ⚠️ Important distinction: When the remote ring back mode is set to 183 Session Progress + SDP, this timer may apply instead of SS_SIP_TIMEOUT_RINGING. Understanding which timer governs your call depends on the ring back mode configured for the gateway. For a deeper understanding of how these SIP sessions work, see our VOS3000 SIP session guide. 🔗
🔔 The SS_SIP_TIMEOUT_RINGING parameter defines the maximum number of seconds a call will remain in the “ringing” or “alerting” state before VOS3000 terminates the call attempt. When VOS3000 sends a SIP INVITE and receives a 180 Ringing response, the ringing timer starts counting. If the called party does not answer within the configured timeout, VOS3000 sends a CANCEL or BYE to end the call attempt. 📞
| Attribute | Value |
|---|---|
| 📌 Parameter Name | SS_SIP_TIMEOUT_RINGING |
| 🔢 Default Value | 120 |
| 📐 Unit | Seconds |
| 📝 Description | SIP Ringing timeout. Default value in “Routing Gateway > Additional settings > Protocol > SIP” |
| 📍 Navigation | Operation management → Softswitch management → Additional settings → SIP parameter |
💡 Key insight: The default of 120 seconds (2 minutes) means that if a called party does not pick up within 2 minutes of ringing, VOS3000 will automatically terminate the call. This is a reasonable default for most deployments, but your specific use case may require a different value — especially when no-answer call forwarding is involved.
🎯 One of the most critical implications of the VOS3000 SIP ringing timeout is its direct relationship with no-answer call forwarding. When a call hits the ringing timeout and is classified as “no answer,” VOS3000 can automatically forward the call to an alternate destination — but only if the ringing timeout has been configured to allow enough time for the original destination to answer. ⚙️
| Ringing Timeout | No-Answer Forward | Total Caller Wait | Use Case |
|---|---|---|---|
| 15s | Yes — after 15s | 15s + forward ringing | 📞 Quick mobile forwarding |
| 30s | Yes — after 30s | 30s + forward ringing | 🏢 PBX extension forwarding |
| 60s | Yes — after 60s | 60s + forward ringing | 🔧 Patient desk phone ring |
| 120s (default) | Yes — after 120s | 120s + forward ringing | ⚠️ Long wait — may frustrate callers |
💡 Recommendation: If you are using no-answer call forwarding, set the VOS3000 SIP ringing timeout to 30-45 seconds for mobile destinations and 45-60 seconds for desk phones. The default 120 seconds is too long for most forwarding scenarios — callers will hang up before the forward triggers. 📱
🖥️ VOS3000 also provides a separate ringing timeout for IVR scenarios. The IVR_RINGING_TIMEOUT parameter controls how long IVR will ring before hanging up when there is no reply. 🔔
| Attribute | Value |
|---|---|
| 📌 Parameter Name | IVR_RINGING_TIMEOUT |
| 🔢 Default Value | 120 |
| 📐 Unit | Seconds |
| 📝 Description | Time for IVR Hang Up, When No Reply |
🎯 Key difference: While SS_SIP_TIMEOUT_RINGING governs the SIP signaling timeout for all calls, IVR_RINGING_TIMEOUT specifically controls IVR-directed call scenarios. If your IVR transfers calls to agents and the agents do not answer, this timer determines when the IVR gives up. For call center deployments, you may want to set this to 30-45 seconds to ensure callers are not stuck listening to endless ringing before being returned to queue or voicemail. 📞
🤔 A common source of confusion in VOS3000 deployments is the distinction between 100 Trying, 183 Session Progress, and 180 Ringing. All are SIP provisional (1xx) responses, but they serve very different purposes in the call setup signal chain and trigger different timers: 📊
| Aspect | 100 Trying | 183 Session Progress (no SDP) | 183 Session Progress (with SDP) | 180 Ringing |
|---|---|---|---|---|
| 📌 SIP Code | 100 | 183 | 183 | 180 |
| 📡 Meaning | Request received, processing | Call is being progressed | Call progressing + media established | Destination is ringing |
| 🎵 Media Path | No | No | Yes — early media | No (local ringback) |
| 🔄 Forwarded downstream? | No — hop-by-hop only | Yes — end-to-end | Yes — end-to-end | Yes — end-to-end |
| ⏱️ VOS3000 Timeout | SS_SIP_TIMEOUT_TRYING (20s) | SS_SIP_TIMEOUT_SESSION_PROGRESS (20s) | SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s) | SS_SIP_TIMEOUT_RINGING (120s) |
| 🎯 Typical use case | Gateway received INVITE, searching route | Call routing in progress, hold on | Playing IVR, queue announcement, ringback | Destination phone is alerting |
🔧 VOS3000 allows you to override all four VOS3000 SIP call progress timeout values on a per-gateway basis. This is configured in the Routing Gateway > Additional settings > Protocol > SIP section for each gateway. 💡
📊 Why override per gateway? Different termination providers and gateway types behave very differently during call setup:
| Gateway Setting | Global Default Source | Description |
|---|---|---|
| Trying timeout | SS_SIP_TIMEOUT_TRYING (20s) | Overrides how long to wait after 100 Trying |
| SessionProgress(183) timeout | SS_SIP_TIMEOUT_SESSION_PROGRESS (20s) | Overrides 183 without SDP timeout |
| SessionProgress(SDP) timeout | SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s) | Overrides 183 with SDP / early media timeout |
| Ringing timeout | SS_SIP_TIMEOUT_RINGING (120s) | Overrides ringing timeout for this gateway |
| Remote ring back mode | Gateway-specific | Controls how ringback is delivered to the caller |
⚙️ This per-gateway granularity is powerful. You can give a slow international carrier 30 seconds of trying timeout while keeping fast domestic gateways at the default 20 seconds. For help with gateway configuration, see our gateway configuration and routing mapping guide. 🔗
🔔 The Remote ring back mode setting in each gateway’s SIP configuration determines how VOS3000 handles the alerting signal sent back to the caller. This directly interacts with the VOS3000 SIP call progress timeout behavior. 🎯
| Mode | SIP Response | Behavior | Active Timer |
|---|---|---|---|
| 🔔 Passthrough | 180 or 183 as received | Forwards the remote party’s response unchanged | Ringing or Session Progress (based on response) |
| 📞 183 Session Progress + SDP | 183 with SDP body | VOS3000 generates 183 with SDP for early media | SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s) |
| 📱 180 Alerting + SDP | 180 with SDP body | VOS3000 generates 180 with SDP for ringback tone | SS_SIP_TIMEOUT_RINGING (120s) |
⚠️ Important distinction: When the remote ring back mode is set to 183 Session Progress + SDP, the call enters early media state. In this case, SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (default: 120 seconds) applies instead of SS_SIP_TIMEOUT_RINGING. Understanding which timer governs your call depends on the ring back mode configured for the gateway. For detailed information on how these SIP responses flow through your softswitch, refer to our VOS3000 SIP session guide. 🔧
⚙️ Follow these steps to configure all four signal progress timeout parameters on your VOS3000 system: 📋
📝 After configuration, verify the timeouts are working correctly using SIP debug tools. For comprehensive debugging instructions, see our VOS3000 SIP debug guide. 🔎
🎯 Different VoIP deployment scenarios require different signal progress timeout values. Here are our recommended settings based on real-world experience: 💡
| Deployment Type | Trying | 183 Timeout | 183 SDP Timeout | Ringing |
|---|---|---|---|---|
| 📞 Mobile termination | 20s | 15s | 60s | 30-45s |
| 🏢 Enterprise PBX | 20s | 20s | 120s | 45-60s |
| 🌍 International routes | 30s | 25s | 90s | 60s |
| 🔔 IVR / Call center | 20s | 15s | 90s | 20-30s |
| 📡 SIP trunking | 20s | 20s | 120s | 60-90s |
| 🛡️ Premium routes | 25s | 20s | 120s | 90-120s |
⚠️ Important note: The VOS3000 SIP call progress timeout must be coordinated with your call routing failover configuration. If the trying timeout is shorter than the time it takes for a backup route to be tried, you may need to adjust either the timeout or the failover strategy. 🔧
❌ Misconfigured call progress timeouts cause a range of frustrating issues. Here are the most common problems and their solutions: 🔍
🔍 Symptom: Calls to specific gateways consistently fail exactly 20 seconds after the INVITE, even though the far end eventually responds.
💡 Cause: The SS_SIP_TIMEOUT_TRYING (20 seconds) is expiring before the gateway can send a progress signal. This is common with international routes that have multiple SIP hops.
✅ Solutions:
🔍 Symptom: Calls where the caller is hearing IVR audio or queue announcements get cut off at 20 seconds.
💡 Cause: The far-end gateway is sending a 183 Session Progress without SDP, so SS_SIP_TIMEOUT_SESSION_PROGRESS (20s) applies instead of SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s). Or the gateway is sending a 100 Trying followed by silence, triggering the trying timeout.
✅ Solutions:
🔍 Symptom: Gateway channels fill up with unanswered calls, new calls fail with “no available channels.”
💡 Cause: SS_SIP_TIMEOUT_RINGING is set too high (or using the default 120s for mobile routes).
✅ Solutions:
🔍 Symptom: Some early media calls time out at 20 seconds while others last 120 seconds, even on the same gateway.
💡 Cause: The far end is inconsistently including or omitting the SDP body in 183 responses. When SDP is present, the 120-second timer applies; when absent, the 20-second timer fires. This is common when multiple upstream providers are reached through the same gateway.
✅ Solutions:
🔍 Symptom: Calls are forwarded on no-answer inconsistently or not at all.
💡 Cause: The caller hangs up before the ringing timeout expires, so the “no-answer” condition is never reached — instead, it is recorded as a “caller hangup.”
✅ Solutions:
✅ Use this checklist when deploying or tuning your VOS3000 SIP call progress timeout settings: 📋
| Check | Action | Status |
|---|---|---|
| 📌 1 | Set SS_SIP_TIMEOUT_TRYING (default: 20s) based on gateway response times | ☐ |
| 📌 2 | Set SS_SIP_TIMEOUT_SESSION_PROGRESS (default: 20s) based on gateway behavior | ☐ |
| 📌 3 | Set SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (default: 120s) to match IVR/queue hold times | ☐ |
| 📌 4 | Set SS_SIP_TIMEOUT_RINGING (default: 120s) to appropriate value for your deployment | ☐ |
| 📌 5 | Configure per-gateway overrides for slow international routes | ☐ |
| 📌 6 | Set Remote ring back mode for each gateway (Passthrough / 183 + SDP / 180 + SDP) | ☐ |
| 📌 7 | Configure IVR_RINGING_TIMEOUT for call center scenarios | ☐ |
| 📌 8 | Verify with SIP debug to confirm correct timer fires at correct interval | ☐ |
| 📌 9 | Check CDR records for call end reasons to verify timeout classification | ☐ |
| 📌 10 | Coordinate no-answer call forwarding timing with ringing timeout | ☐ |
⏱️ The VOS3000 SIP call progress timeout chain is a sequence of four timers that fire during the SIP call setup process: SS_SIP_TIMEOUT_TRYING (20s, triggered by 100 Trying), SS_SIP_TIMEOUT_SESSION_PROGRESS (20s, triggered by 183 without SDP), SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (120s, triggered by 183 with SDP), and SS_SIP_TIMEOUT_RINGING (120s, triggered by 180 Ringing). Each timer monitors a specific stage of call progress and hands off to the next when the call advances. If any timer expires without progress, the call is terminated. 📡
📊 The difference depends on which SIP response the gateway sends. If the gateway sends a 100 Trying or 183 Session Progress without SDP, the 20-second timer applies because no media is flowing. If the gateway sends a 183 Session Progress with SDP or a 180 Ringing, the 120-second timer applies because the call is in an active state (early media or alerting). Check your gateway’s Remote ring back mode setting and inspect the SIP trace to see which responses contain SDP. 🔧
🖥️ Yes! VOS3000 supports per-gateway overrides for all four call progress timeout parameters. Navigate to Routing Gateway > [Select Gateway] > Additional settings > Protocol > SIP and set the individual timeout fields. If left blank, the gateway uses the global default. This is especially useful when you have both mobile and fixed-line gateways that require different timeout values. 🔧
🔄 When the VOS3000 SIP ringing timeout expires, the call is classified as “no-answer” and terminated. If no-answer call forwarding is configured, VOS3000 forwards the call at this point. This means the ringing timeout directly determines when the forwarding triggers. Set it too long and the caller hangs up first; set it too short and legitimate answers are missed. A recommended range is 30-45 seconds for mobile destinations with forwarding enabled. 📞
📊 SS_SIP_TIMEOUT_RINGING (default: 120s) applies when VOS3000 receives a 180 Ringing response. SS_SIP_TIMEOUT_SESSION_PROGRESS_SDP (default: 120s) applies when VOS3000 receives a 183 Session Progress with SDP, which establishes early media. Which timer applies depends on the gateway’s Remote ring back mode setting and the actual SIP response from the far end. Both default to 120 seconds but can be configured independently. 📡
🔍 Start by capturing a SIP trace using the methods described in our SIP debug guide. Look for the timing between provisional responses and identify which timer is firing. Verify the actual timeout matches your configured value. Check CDR records for the call end reason codes. If calls are timing out at 20 seconds instead of your configured value, check whether the gateway is using 183 Session Progress mode (which triggers SS_SIP_TIMEOUT_SESSION_PROGRESS instead). For complex issues, contact us on WhatsApp at +8801911119966 for expert support. 📞
🔧 Configuring the VOS3000 SIP call progress timeout chain correctly is essential for optimizing your VoIP network’s channel utilization, caller experience, and call forwarding behavior. Whether you need help with global parameter tuning, per-gateway overrides, or troubleshooting timeout-related call failures, our team is ready to assist. 🛡️
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