Test Your Network for
SIP Phone Compatibility
Check if your home or office network can reach SIP port 5060 for VoIP calls. This tests your network's ability to connect to SIP servers for phone registration.
Tests outbound connectivity from your network to port 5060
Port 5060 Check
Tests if your network allows outbound SIP connections
Latency & Jitter
Measures connection quality for voice calls
Recommendations
Get actionable steps to fix issues
Understanding SIP Phone Requirements
What is SIP?
SIP (Session Initiation Protocol) is the standard protocol used for VoIP phone calls. It handles call setup, management, and teardown between endpoints. For SIP to work properly, your network must allow traffic on port 5060 (UDP/TCP) and have sufficient quality for real-time voice communication.
This test verifies that your home or office network can support SIP phones reliably.
Why Test Your Network?
Many home networks have firewalls or routers that block SIP traffic by default. Additionally, poor network quality (high latency, jitter, or packet loss) can cause:
- • Choppy or robotic-sounding audio
- • Dropped calls or connection failures
- • One-way audio (you can't hear them or vice versa)
- • Delayed audio (talking over each other)
Network Quality Requirements
Latency
Round-trip time for voice packets
Jitter
Variation in packet arrival time
Packet Loss
Percentage of lost voice packets
Common Issues & Solutions
Port 5060 Blocked
Solution: Log into your router and forward UDP port 5060 to your computer. You may also need to disable SIP ALG (Application Layer Gateway) which can interfere with SIP traffic.
High Latency or Jitter
Solution: Use a wired Ethernet connection instead of WiFi. Enable QoS (Quality of Service) on your router to prioritize VoIP traffic. Close bandwidth-intensive applications during calls.
Symmetric NAT
Solution: Some routers use Symmetric NAT which can cause SIP issues. Try enabling STUN support in your phone system, or consider upgrading to a business-grade router.