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
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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