VOS3000 Media Proxy and System Parameters: Complete Configuration Reference
VOS3000 media proxy and system parameters control the core functionality of your VoIP softswitch. Proper configuration of these parameters determines call quality, NAT traversal success, security levels, and overall system performance. This comprehensive reference guide covers all critical parameters from the official VOS3000 2.1.9.07 manual, explaining their functions and recommended configurations for different deployment scenarios.
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Table of Contents
📡 Understanding Media Proxy in VOS3000
Media proxy determines whether RTP (Real-time Transport Protocol) voice packets flow directly between endpoints or through the VOS3000 server. This decision has significant implications for NAT traversal, audio quality, server resource usage, and call reliability.
📊 VOS3000 Media Proxy Modes
The SS_MEDIAPROXYMODE parameter controls media proxy behavior with four distinct modes:
| Mode | Behavior | Server Load | Best Use Case |
|---|---|---|---|
| Off | Never proxy media; RTP flows directly between endpoints | Lowest | Public IP endpoints, no NAT issues |
| On | Always proxy all media through server | Highest | Troubleshooting, maximum control |
| Auto | Intelligent decision based on conditions | Variable | Mixed environments, recommended |
| Must On | Forced proxy regardless of other settings | Highest | Specific debugging scenarios only |
⚙️ Media Proxy Auto Mode Decision Logic (VOS3000 Media Proxy)
When SS_MEDIAPROXYMODE is set to “Auto,” VOS3000 follows a precise decision algorithm to determine whether media proxy is needed:
Media Proxy Decision Steps (Auto Mode):
Step 1: Check if caller or callee MUST have media proxy
├── If gateway/phone has Media Proxy = Must On
└── Result: ENABLE media proxy
Step 2: Check if caller or callee has Media Proxy disabled
├── If gateway/phone has Media Proxy = Off
└── Result: DISABLE media proxy
Step 3: Check if caller or callee has Media Proxy enabled
├── If gateway/phone has Media Proxy = On
└── Result: ENABLE media proxy
Step 4: Check if callee has local ring enabled
├── Local ring requires media proxy for ringback tone
└── Result: ENABLE media proxy
Step 5: Check for dynamic registration with encryption
├── If phone/gateway uses dynamic register AND encryption
└── Result: ENABLE media proxy
Step 6: Check cross-network routing (SS_MEDIAPROXYBETWEENNET)
├── If caller and callee from different networks
└── Result: ENABLE media proxy
Step 7: Check NAT conditions (SS_MEDIAPROXYBEHINDNAT)
├── If phone and gateway in same NAT, SS_MEDIAPROXYSAMENAT = On
├── If phone and gateway in different NAT, one in private network
└── Result: ENABLE media proxy
Step 8: Default action
└── Result: DISABLE media proxy
🔧 Configuring Media Proxy Parameters
📍 Location in VOS3000 Client
Navigation Path: Operation Management → Softswitch Management → Additional Settings → System Parameter Parameter Name: SS_MEDIAPROXYMODE Valid Values: Off, On, Auto, Must On Default Value: Auto Related Parameters: ┌─────────────────────────────────────────────────────────────┐ │ Parameter Name │ Description │ ├─────────────────────────────────────────────────────────────┤ │ SS_MEDIAPROXYBETWEENNET │ Proxy for cross-network │ │ SS_MEDIAPROXYBEHINDNAT │ Proxy for behind-NAT │ │ SS_MEDIAPROXYSAMENAT │ Proxy for same-NAT │ └─────────────────────────────────────────────────────────────┘
📡 RTP Port Configuration (VOS3000 Media Proxy)
RTP port configuration determines which UDP ports VOS3000 uses for voice media streams. Proper configuration is essential for firewall rules and capacity planning. VOS3000 Media Proxy
📊 RTP Port Parameters VOS3000 Media Proxy
| Parameter | Default Value | Description |
|---|---|---|
| SS_RTP_PORT_RANGE | 10000,39999 | UDP port range for RTP media streams |
| SS_H245_PORT_RANGE | 10000,39999 | H.245 port range for H.323 calls |
| IVR_RTP_PORT | 40000,47999 | RTP port range for IVR services |
⚙️ RTP Port Sizing Calculation
RTP Port Capacity Planning: Each concurrent call uses 2 RTP ports (one for each direction) Port Range: 10000-39999 = 30,000 ports Maximum Concurrent Calls = 30,000 / 2 = 15,000 calls However, consider: - Each port allocation has overhead - IVR services need separate port range - H.323 calls share same range Recommended Configuration by Capacity: ┌──────────────────────────────────────────────────────────────┐ │ Expected Capacity │ RTP Port Range │ IVR Port Range │ ├──────────────────────────────────────────────────────────────┤ │ Small (<500 CC) │ 10000-19999 │ 40000-40999 │ │ Medium (500-2000) │ 10000-29999 │ 40000-41999 │ │ Large (2000-5000) │ 10000-39999 │ 40000-44999 │ │ Enterprise (5000+)│ 10000-59999 │ 60000-64999 │ └──────────────────────────────────────────────────────────────┘ Firewall Rule Example: iptables -A INPUT -p udp --dport 10000:39999 -j ACCEPT iptables -A INPUT -p udp --dport 40000:47999 -j ACCEPT
🔑 SIP Parameters Reference – VOS3000 Media Proxy
SIP parameters control how VOS3000 handles SIP signaling, authentication, and session management. These parameters directly impact call setup success and session reliability.
📊 Critical SIP Parameters
| Parameter | Default | Purpose |
|---|---|---|
| SS_SIP_NAT_KEEP_ALIVE_MESSAGE | HELLO | Content of NAT keep-alive message |
| SS_SIP_NAT_KEEP_ALIVE_PERIOD | 30 | Keep-alive interval in seconds (10-86400) |
| SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL | 500 | Interval between sending keep-alives (ms) |
| SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME | 3000 | Number of keep-alives sent per batch |
| SS_SIP_SESSION_TTL | 1800 | Session Timer TTL in seconds |
| SS_SIP_SESSION_UPDATE_SEGMENT | 300 | Session update interval in seconds |
| SS_SIP_RESEND_INTERVAL | 0.5,1,2,4,4,4,4,4,4,4 | SIP message resend intervals (seconds) |
| SS_SIP_NO_TIMER_REINVITE_INTERVAL | 7200 | Max call time for non-timer SIP clients |
⚙️ NAT Keep-Alive Configuration
NAT Keep-Alive Purpose: - Maintains NAT binding for devices behind NAT - Prevents one-way audio caused by expired bindings - Essential for devices that don't support SIP Timer How It Works: 1. VOS3000 sends UDP message to registered device IP 2. Message content = SS_SIP_NAT_KEEP_ALIVE_MESSAGE (default: "HELLO") 3. Sent every SS_SIP_NAT_KEEP_ALIVE_PERIOD seconds (default: 30) 4. This keeps the NAT mapping active Configuration Example: SS_SIP_NAT_KEEP_ALIVE_MESSAGE = "HELLO" SS_SIP_NAT_KEEP_ALIVE_PERIOD = 30 SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL = 500 SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME = 3000 This means: - Send "HELLO" to each device every 30 seconds - Wait 500ms between sending to different devices - Process 3000 devices in each batch Scaling Notes: - 3000 devices × 500ms = 25 minutes to process all - Adjust SEND_ONE_TIME for large deployments - Increase SEND_INTERVAL if network is slow
🔐 Authentication Parameters
Authentication parameters control how VOS3000 handles SIP authentication challenges and account lockout policies for security.
📊 Authentication Security Parameters
| Parameter | Default | Purpose |
|---|---|---|
| SS_AUTHENTICATION_MAX_RETRY | 6 | Max auth retries before suspension (0-999) |
| SS_AUTHENTICATION_FAILED_SUSPEND | 180 | Suspension duration in seconds (60-3600) |
| SS_SIP_AUTHENTICATION_CODE | Unauthorized(401) | SIP response code for auth challenge |
| SS_SIP_AUTHENTICATION_TIMEOUT | 10 | Timeout for SIP authentication in seconds |
| SS_SIP_AUTHENTICATION_RETRY | 6 | SIP auth retry count for 401/407 responses |
⚙️ Authentication Lockout Configuration
Security Configuration Example: For High-Security Environments: SS_AUTHENTICATION_MAX_RETRY = 3 SS_AUTHENTICATION_FAILED_SUSPEND = 300 For Standard Environments: SS_AUTHENTICATION_MAX_RETRY = 6 SS_AUTHENTICATION_FAILED_SUSPEND = 180 For Relaxed Environments (trusted networks only): SS_AUTHENTICATION_MAX_RETRY = 10 SS_AUTHENTICATION_FAILED_SUSPEND = 60 How Lockout Works: 1. Device attempts registration with wrong password 2. VOS3000 returns 401 Unauthorized 3. Device retries (up to SS_AUTHENTICATION_MAX_RETRY times) 4. After max retries, IP is added to temporary block list 5. Block lasts for SS_AUTHENTICATION_FAILED_SUSPEND seconds 6. After timeout, device can retry This protects against: - Brute force password attacks - SIP flood attacks - Credential guessing - Automated hacking tools
📊 Session Timer Configuration (VOS3000 Media Proxy)
Session timers ensure that hung calls are detected and cleaned up, preventing “ghost calls” and billing errors.
⚙️ Session Timer Parameters
Session Timer Configuration: SS_SIP_SESSION_TTL = 1800 (30 minutes) SS_SIP_SESSION_UPDATE_SEGMENT = 300 (5 minutes) SS_SIP_NO_TIMER_REINVITE_INTERVAL = 7200 (2 hours) How SIP Session Timer Works: 1. During call setup, session timer is negotiated 2. VOS3000 sends UPDATE or re-INVITE at interval 3. If no response, session is considered dead 4. Call is terminated and CDR is generated For Non-Timer-Capable Clients: - SS_SIP_NO_TIMER_REINVITE_INTERVAL sets max call time - After this duration, call is terminated - Prevents ultra-long "zombie" calls Recommended Values: ┌────────────────────────────────────────────────────────────┐ │ Scenario │ TTL │ Update Segment │ Max No-Timer │ ├────────────────────────────────────────────────────────────┤ │ Standard VoIP │ 1800 │ 300 │ 7200 │ │ High-Volume Trunk │ 3600 │ 600 │ 14400 │ │ Calling Card │ 900 │ 180 │ 3600 │ │ Enterprise PBX │ 1800 │ 300 │ 28800 │ └────────────────────────────────────────────────────────────┘ Session Timer Benefits: - Detects hung calls automatically - Prevents billing discrepancies - Reduces "ghost call" complaints - Frees system resources
🎯 H.323 Parameters Reference
For environments using H.323 protocol, VOS3000 provides comprehensive parameter controls.
📊 Critical H.323 Parameters
| Parameter | Default | Purpose |
|---|---|---|
| SS_H245_PORT_RANGE | 10000,39999 | Port range for H.245 control channel |
| SS_H323_DTMF_METHOD | H.245 alphanumeric | Default DTMF transmission method |
| SS_H323_TIMEOUT_ALERTING | 120 | Timeout for alerting state (seconds) |
| SS_H323_TIMEOUT_CALLPROCEEDING | 20 | Timeout for call proceeding (seconds) |
| SS_H323_TIMEOUT_SETUP | 5 | Timeout for call setup (seconds) |
📈 Quality of Service (QoS) Parameters
QoS parameters control the DSCP marking on IP packets for prioritization in managed networks.
⚙️ QoS Configuration
QoS Parameters: SS_QOS_SIGNAL = 0xa0 (default) - DSCP marking for SIP/H.323 signaling packets - Hex value applied to IP header ToS field SS_QOS_RTP = 0xa0 (default) - DSCP marking for RTP media packets - Hex value applied to IP header ToS field DSCP Value Reference: ┌─────────────────────────────────────────────────────────────┐ │ Hex Value │ Binary │ DSCP Class │ Description │ ├─────────────────────────────────────────────────────────────┤ │ 0x00 │ 000000 │ Best Effort │ Default, no QoS │ │ 0x20 │ 001000 │ CS1 │ Scavenger │ │ 0x40 │ 010000 │ CS2 │ OAM │ │ 0x60 │ 011000 │ CS3 │ Signaling │ │ 0x80 │ 100000 │ CS4 │ Real-time │ │ 0xa0 │ 101000 │ CS5 / EF │ Voice (default) │ │ 0xc0 │ 110000 │ CS6 │ Network control │ └─────────────────────────────────────────────────────────────┘ When to Configure: - Only in managed networks with QoS policies - Coordinate with network team on DSCP values - Match router/switch QoS configuration
📊 Billing and CDR Parameters
These parameters control billing precision and CDR generation behavior. VOS3000 Media Proxy
⚙️ Critical Billing Parameters
| Parameter | Default | Purpose |
|---|---|---|
| SERVER_BILLING_HOLD_TIME_PRECISION | 50 | Billing time precision in milliseconds |
| SERVER_MAX_CDR_PENDING_LIST_LENGTH | 100000 | Max pending CDR queue length |
| SERVER_CDR_FILE_WRITE_MAX | 2048 | Max CDR files to retain |
| SERVER_CDR_FILE_WRITE_INTERVAL | 60 | CDR file write interval (seconds) |
🔗 Related Resources – VOS3000 Media Proxy
- 📖 VOS3000 Extended Firewall Configuration
- 📖 VOS3000 One-Way Audio Troubleshooting
- 📖 VOS3000 Server Configuration Guide
- 📖 VOS3000 FAQ Based on Official Manual
- 📖 VOS3000 Downloads
❓ Frequently Asked Questions
Should I set media proxy to On or Auto?
Auto mode is recommended for most deployments. It intelligently enables media proxy only when needed (NAT traversal, encryption, cross-network calls) while allowing direct RTP when possible. This provides the best balance of reliability and server resource usage.
How do I know if my RTP port range is sufficient?
Calculate: Each concurrent call uses 2 RTP ports. With default range 10000-39999 (30,000 ports), you can support 15,000 concurrent calls. Monitor port usage through system performance monitoring. If you see port allocation errors, increase the range or reduce concurrent call load.
Why do calls drop at 30 seconds?
This typically indicates SIP session timer or NAT binding issues. Check SS_SIP_SESSION_TTL and ensure NAT keep-alive is configured. The 30-second timeout often corresponds to NAT binding expiry when keep-alives are not working.
What is the best authentication retry setting?
For most environments, the default of 6 retries with 180-second suspension works well. For high-security environments, reduce to 3 retries with longer suspension (300+ seconds). Balance security against false positives from legitimate users mistyping passwords.
How do I troubleshoot media proxy issues?
Use Debug Trace in VOS3000 to capture SIP and SDP messages. Check if media proxy is being invoked (look at the c= line in SDP). Verify that RTP ports are within configured range. Check firewall rules allow both signaling and RTP ports.
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