SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons

VOS3000 Media Proxy and System Parameters: Complete Important Configuration Reference

March 31, 2026March 31, 2026 king

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.

📞 Need help configuring VOS3000 parameters? WhatsApp: +8801911119966

📡 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)

📖 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.

📞 Get Expert Help with VOS3000 Configuration

Need assistance optimizing VOS3000 parameters for your specific deployment? Our team provides professional VOS3000 installation, configuration, and performance tuning services.

📱 WhatsApp: +8801911119966

📞 Need Professional VOS3000 Setup Support?

For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:

📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads

VOS3000 softswitch VoIP, VOS3000 seguridad, VOS3000 Call Center Soluciones, VOS3000 API Integración, VOS3000 Infraestructura, VOS3000 Errores Ruting Llamadas, VOS3000错误代码替换与呼叫失败排查, VOS3000 Optimización de Rendimiento, VOS3000 Códigos Error Terminación, VOS3000 NoAvailableRouter错误解决方案, Negocio VoIP Mayorista, VICIDIAL Servidor, Softswitch Barato, VoIP批发业务, 软交换比较, Advance Routing, VOS3000 Troubleshooting Guide, VOS3000 CDR Analysis, Guía Completa VOS3000 2026, VOS3000 指南 2026, SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons

SIP ALG Problems: Complete Troubleshooting Guide for VoIP NAT Issues

March 31, 2026March 31, 2026 king

SIP ALG Problems: Complete Troubleshooting Guide for VoIP NAT Issues

SIP ALG problems are among the most frustrating issues facing VoIP administrators and telecom operators today. When SIP Application Layer Gateway (ALG) functionality interferes with VoIP traffic, it causes registration failures, one-way audio, dropped calls, and complete communication breakdowns. This comprehensive troubleshooting guide covers everything you need to know about diagnosing and resolving SIP ALG problems across all major router brands and network configurations.

📞 Need help with VoIP NAT issues? WhatsApp: +8801911119966

🔍 What is SIP ALG and Why Does It Break VoIP?

SIP ALG (Application Layer Gateway) is a feature implemented in many routers and firewalls that is designed to help SIP traffic traverse NAT (Network Address Translation) boundaries. While the intention is good, SIP ALG implementations are notoriously problematic and often cause more harm than benefit for VoIP deployments.

📊 How SIP ALG Works (In Theory)

The SIP ALG function monitors SIP signaling traffic and attempts to modify SIP headers and SDP (Session Description Protocol) payloads to help with NAT traversal. When a SIP message passes through a NAT device, the ALG examines the packet and rewrites IP addresses and port numbers to match the public-facing NAT address instead of the private internal address.

❌ Why SIP ALG Causes Problems (SIP ALG Problems)

Problem Type	How SIP ALG Causes It	Technical Explanation
One-Way Audio	Incorrect SDP modification	ALG rewrites SDP to wrong IP/port, causing RTP to fail in one direction
Registration Failures	Contact header corruption	ALG modifies Contact header incorrectly, server cannot reach client
Call Drops at 30 Seconds	SIP timer interference	ALG interferes with SIP keep-alive and session timers
No Incoming Calls	NAT binding loss	ALG-created bindings expire prematurely, incoming INVITE fails
Duplicate SIP Messages	Packet replication	Poorly implemented ALG creates duplicate packets causing confusion

🚨 Common SIP ALG Problem Symptoms (SIP ALG Problems)

Identifying SIP ALG problems requires recognizing specific patterns in VoIP behavior. The following symptoms strongly indicate SIP ALG interference in your network:

📋 Symptom Checklist

One-Way Audio: Call connects but only one party can hear audio, typically the internal party cannot hear external caller
No Audio on Answer: Phone rings and answers, but complete silence on both ends
Registration Expiry: Extensions register initially but lose registration within minutes
30-Second Call Drops: Calls disconnect precisely at 30-second intervals due to NAT binding timeout
Incoming Call Failures: Outbound calls work fine but inbound calls never reach the phone
Intermittent Issues: Problems appear and disappear without apparent pattern
VPN vs Direct: VoIP works through VPN but fails on direct internet connection

🔧 How to Disable SIP ALG on Popular Routers

Disabling SIP ALG is often the most direct solution to VoIP NAT problems. Below are instructions for major router brands commonly found in VoIP deployments:

🔷 Cisco Routers

On Cisco IOS routers, SIP ALG is implemented as SIP inspection in the firewall configuration:

! Check current SIP inspection status
show running-config | include sip

! Disable SIP inspection in class-map
configure terminal
class-map inspection_default
  no match protocol sip

! Or remove from policy-map
policy-map global_policy
  class inspection_default
    no inspect sip

! Save configuration
write memory

🔷 Fortinet FortiGate

FortiGate firewalls have SIP ALG enabled by default. Disable through CLI or GUI:

! Via CLI - Check SIP helper status
diagnose sys sip-proxy status

! Disable SIP helper
config system settings
  set sip-helper disable
  set sip-nat-trace disable
end

! Also check VOIP profile
config voip profile
  edit default
    config sip
      set status disable
    end
  next
end

🔷 MikroTik RouterOS

MikroTik routers use SIP helper for ALG functionality:

# Check SIP helper status
/ip firewall service-port print

# Disable SIP helper
/ip firewall service-port disable sip

# For older RouterOS versions
/ip firewall nat disable [find comment="SIP"]

🔷 TP-Link Routers

TP-Link consumer and business routers have SIP ALG in different locations:

TP-Link Model	Menu Location	Setting
Archer Series	Advanced → NAT Forwarding → ALG	Uncheck “SIP ALG”
TL-ER Series	Network → ALG	Disable SIP checkbox
Omada SDN	Settings → Transmission → NAT	Toggle SIP ALG off

🔷 Netgear Routers

# Web Interface Navigation
# 1. Login to router admin panel
# 2. Go to Advanced → Setup → WAN Setup
# 3. Find "SIP ALG" or "SIP Connection Tracking"
# 4. Uncheck/disable the option
# 5. Apply changes and reboot router

🔷 Asus Routers

# Web Interface
# 1. Advanced Settings → WAN
# 2. NAT Passthrough tab
# 3. Set "SIP Passthrough" to "Disable"
# 4. Apply and reboot

# Via SSH/Telnet
nvram set sip_passthrough=0
nvram commit
reboot

🔷 Ubiquiti UniFi / EdgeRouter

# UniFi Security Gateway
# Via config.gateway.json:
{
  "service": {
    "nat": {
      "rule": {
        "5000": {
          "description": "Disable SIP ALG",
          "log": "disable",
          "protocol": "all",
          "source": {
            "group": {
              "network-group": "net_LAN"
            }
          },
          "type": "masquerade"
        }
      }
    }
  }
}

# EdgeRouter CLI
configure
set service nat rule 5000 disable
commit
save

🌐 NAT Traversal Solutions Beyond Disabling SIP ALG (SIP ALG Problems)

In some network environments, simply disabling SIP ALG is not sufficient or may not be possible. Understanding and implementing proper NAT traversal techniques ensures reliable VoIP operation.

📊 NAT Traversal Methods Comparison

Method	How It Works	Pros	Cons
STUN Server	Client discovers public IP/port	Simple, low overhead	Does not work with symmetric NAT
TURN Server	Media relayed through server	Works with all NAT types	Higher latency, server load
ICE Protocol	Tries STUN first, falls back to TURN	Best of both methods	More complex configuration
Media Proxy	Server proxies RTP traffic	Server controls media path	Additional server resources

📡 VOS3000 NAT Configuration

For VOS3000 softswitch deployments, proper NAT configuration is essential. VOS3000 provides several parameters to handle NAT traversal scenarios:

Parameter	Default	Purpose
SS_SIP_NAT_KEEP_ALIVE_MESSAGE	HELLO	Content of NAT keep-alive message sent to maintain NAT bindings
SS_SIP_NAT_KEEP_ALIVE_PERIOD	30	Interval in seconds between NAT keep-alive messages (10-86400)
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL	500	Interval in milliseconds between sending keep-alives to different devices
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME	3000	Number of NAT keep-alive messages sent in one batch

🔧 VOS3000 Media Proxy Configuration

VOS3000 supports multiple media proxy modes to handle NAT scenarios. The SS_MEDIAPROXYMODE parameter controls this behavior:

Media Proxy Modes in VOS3000:

ON       - Media proxy always enabled
          All RTP flows through VOS3000 server
          Highest server resource usage

OFF      - Media proxy always disabled
          RTP flows directly between endpoints
          May fail with NAT issues

AUTO     - VOS3000 decides based on conditions:
          1. If caller/callee requires media proxy → Enable
          2. If caller/callee disabled media proxy → Disable
          3. If encryption enabled → Enable
          4. If different networks (SS_MEDIAPROXYBETWEENNET) → Enable
          5. If behind NAT (SS_MEDIAPROXYBEHINDNAT) → Enable
          6. Otherwise → Disable

MUST ON  - Forced media proxy regardless of settings
          Used for specific troubleshooting scenarios

🔍 Diagnosing SIP ALG Problems

📋 Testing for SIP ALG Presence

Before making configuration changes, confirm that SIP ALG is actually causing the problem:

Packet Capture Analysis: Use Wireshark to capture SIP traffic and compare original packets with received packets
Contact Header Check: Look for differences between internal IP and Contact header IP in SIP messages
SDP Analysis: Compare c= (connection) line in SDP with actual endpoint IP
Via Header Inspection: Check if received/rport parameters are being modified incorrectly
Online Tools: Use SIP ALG detection tools available from VoIP providers

📊 Wireshark Filter Commands

# SIP traffic filter
sip

# SIP registration only
sip.Method == "REGISTER"

# SIP invite and responses
sip.Method == "INVITE" || sip.Status-Code

# RTP media streams
rtp

# Check for NAT-related issues
sip.Contact contains "192.168" || sip.Contact contains "10."

📖 VOS3000 Extended Firewall Configuration Guide
📖 VOS3000 One-Way Audio Troubleshooting
📖 VOS3000 Server Configuration Guide
📖 SIP Trunking with VOS3000
📖 VOS3000 Downloads

❓ Frequently Asked Questions

How do I know if my router has SIP ALG enabled?

The most reliable method is to capture SIP traffic using Wireshark and examine the Contact headers and SDP content. If the IP addresses in these fields show your public IP when they should show private IPs (or vice versa), SIP ALG is active. Many router admin interfaces also display SIP ALG status in the NAT or Firewall settings sections.

Will disabling SIP ALG break other applications?

In most cases, disabling SIP ALG does not negatively affect other applications. SIP ALG is specifically designed for SIP protocol and has no impact on web browsing, email, or other network services. However, some legacy SIP devices that rely on ALG for NAT traversal may require alternative NAT configuration after disabling.

Why do calls still drop after disabling SIP ALG?

If problems persist after disabling SIP ALG, other factors may be involved: firewall rules blocking RTP ports, incorrect NAT keep-alive settings, SIP session timer issues, or NAT binding timeouts. Check firewall rules for ports 5060 (SIP) and 10000-20000 (RTP), and verify SIP registration expiry settings.

Can SIP ALG be disabled on ISP-provided routers?

Many ISP-provided routers do not allow SIP ALG configuration through the web interface. Options include: contacting ISP to disable the feature, using bridge mode with a separate router, or replacing the ISP router entirely with a commercial router that offers full configuration access.

What is the difference between SIP ALG and SIP Helper?

SIP ALG and SIP Helper are essentially the same feature with different naming conventions across vendors. Cisco and MikroTik commonly use “SIP Helper,” while Fortinet and others use “SIP ALG.” Both refer to the router’s ability to inspect and modify SIP packets for NAT traversal purposes.

📞 Get Expert Help with SIP ALG Problems

Still experiencing VoIP NAT issues after following this guide? Our team of VoIP experts can help diagnose and resolve SIP ALG problems, configure proper NAT traversal, and optimize your VOS3000 deployment for reliable operation.

📱 WhatsApp: +8801911119966

📞 Need Professional VOS3000 Setup Support?

For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:

📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads

VOS3000 Media Proxy and System Parameters: Complete Configuration Reference

Table of Contents

📡 Understanding Media Proxy in VOS3000

📊 VOS3000 Media Proxy Modes

⚙️ Media Proxy Auto Mode Decision Logic (VOS3000 Media Proxy)

🔧 Configuring Media Proxy Parameters

📍 Location in VOS3000 Client

📡 RTP Port Configuration (VOS3000 Media Proxy)

📊 RTP Port Parameters VOS3000 Media Proxy

⚙️ RTP Port Sizing Calculation

🔑 SIP Parameters Reference – VOS3000 Media Proxy

📊 Critical SIP Parameters

⚙️ NAT Keep-Alive Configuration

🔐 Authentication Parameters

📊 Authentication Security Parameters

⚙️ Authentication Lockout Configuration

📊 Session Timer Configuration (VOS3000 Media Proxy)

⚙️ Session Timer Parameters

🎯 H.323 Parameters Reference

📊 Critical H.323 Parameters

📈 Quality of Service (QoS) Parameters

⚙️ QoS Configuration

📊 Billing and CDR Parameters

⚙️ Critical Billing Parameters

🔗 Related Resources – VOS3000 Media Proxy

❓ Frequently Asked Questions

Should I set media proxy to On or Auto?

How do I know if my RTP port range is sufficient?

Why do calls drop at 30 seconds?

What is the best authentication retry setting?

How do I troubleshoot media proxy issues?

📞 Get Expert Help with VOS3000 Configuration

📞 Need Professional VOS3000 Setup Support?

SIP ALG Problems: Complete Troubleshooting Guide for VoIP NAT Issues

Table of Contents

🔍 What is SIP ALG and Why Does It Break VoIP?

📊 How SIP ALG Works (In Theory)

❌ Why SIP ALG Causes Problems (SIP ALG Problems)

🚨 Common SIP ALG Problem Symptoms (SIP ALG Problems)

📋 Symptom Checklist

🔧 How to Disable SIP ALG on Popular Routers

🔷 Cisco Routers

🔷 Fortinet FortiGate

🔷 MikroTik RouterOS

🔷 TP-Link Routers

🔷 Netgear Routers

🔷 Asus Routers

🔷 Ubiquiti UniFi / EdgeRouter

🌐 NAT Traversal Solutions Beyond Disabling SIP ALG (SIP ALG Problems)

📊 NAT Traversal Methods Comparison

📡 VOS3000 NAT Configuration

⚙️ VOS3000 NAT-Related System Parameters

🔧 VOS3000 Media Proxy Configuration

🔍 Diagnosing SIP ALG Problems

📋 Testing for SIP ALG Presence

📊 Wireshark Filter Commands

🔗 Related Resources (SIP ALG Problems)

❓ Frequently Asked Questions

How do I know if my router has SIP ALG enabled?

Will disabling SIP ALG break other applications?

Why do calls still drop after disabling SIP ALG?

Can SIP ALG be disabled on ISP-provided routers?

What is the difference between SIP ALG and SIP Helper?

📞 Get Expert Help with SIP ALG Problems

📞 Need Professional VOS3000 Setup Support?