Nextiva NEXT - System/Network Requirements
Basic System Requirements
Desktop or Laptop
- Memory: Minimum 8 GB RAM
- Browser: Latest version of Google Chrome or Microsoft Edge (Chromium)
- Audio: Wired USB headset recommended for best call quality
Mobile
- iOS: Version 15 or newer (iPhone 11 or higher recommended)
- Android: Version 10 or newer (mainstream vendors only)
File Upload Support
- Formats: png, jpeg, mp4, mov, mkv, mp3
- Maximum file size: 20 MB
Browser Compatibility
| Browser | Minimum Version | Operating System | Notes |
|---|---|---|---|
| Chrome | 114+ | Windows 10+, macOS 12+ | Recommended |
| Edge (Chromium) | 114+ | Windows 10+ | Full support |
| Firefox | 112+ | Windows 10+, macOS | Limited QA coverage |
| Safari | 15+ | macOS 12+ | Limited QA coverage |
Network Quality and Bandwidth Requirements
| Metric | Requirement | Impact if Poor |
|---|---|---|
| Bandwidth | Minimum 300 kbps per active call; 1–5 Mbps per user recommended | Audio cutouts or dropped calls |
| Packet Loss | < 1% | Choppy or robotic audio |
| Jitter | < 30 ms | Gaps and distortion |
| Latency | < 150 ms | Noticeable voice delay |
Recommendations
- Use a wired Ethernet connection whenever possible.
- If using Wi-Fi, connect to 5 GHz instead of 2.4 GHz.
- Avoid large downloads or video streaming during calls.
Firewall and Router Configuration
FQDN/IP Addresses
| FQDN | IP Address(es) | Purpose | Port Range |
|---|---|---|---|
| sbc-non.sip.nextiva.io | 34.86.166.231, 34.21.116.230 | SIP device signaling |
UDP/TCP 4660–4661, UDP 10000–65535 |
| sbc-uw1-1.sip.nextiva.io | 35.230.30.179, 34.53.32.156 | SIP device signaling | UDP/TCP 4660–4661, UDP 10000–65535 |
| fs-non.webrtc.nextiva.io | 34.54.141.225 | Web/mobile client signaling | TCP 443, UDP 10000–65535 |
| fs-uw1-1.webrtc.nextiva.io | 34.54.141.225 | Web/mobile client signaling | TCP 443, UDP 10000–65535 |
| coturn-e4.webrtc.nextiva.io | 34.86.37.236 | TURN relay (East) | TCP/UDP 443, 5349, UDP 10000–65535 |
| coturn-w1.webrtc.nextiva.io | 34.169.89.130 | TURN relay (West) | TCP/UDP 443, 5349, UDP 10000–65535 |
Tip: Add both the FQDN and IP addresses to your allow-list to ensure reliable media and signaling.
Protocols Used
| Protocol | Port Range | Purpose |
|---|---|---|
| TCP | 443 | WebRTC signaling and secure provisioning |
| TCP/UDP | 443, 3478, 5349 | STUN/TURN NAT traversal |
| UDP | 10000–65535 | RTP/RTCP media traffic |
| TCP/UDP | 4660, 4661 | SIP signaling |
| UDP | 123 | NTP time synchronization |
Firewall Configuration Notes
- Bypass SSL/TLS inspection for all STUN, TURN, and WebRTC signaling domains.
- Allow outbound UDP traffic for dynamic ports (10000–65535).
- Add all FQDNs and IPs listed below to your firewall allow-list.
Additional Settings
To ensure consistent call quality, verify these router settings:
- Disable SIP ALG (also called SIP Helper or ALG Services).
- Avoid Double NAT: Set secondary routers to Bridge or Access Point mode.
- Check WAN IP: Ensure it’s not private (10.x.x.x, 172.16–31.x.x, 192.168.x.x).
Common issues caused by SIP ALG or Double NAT:
- One-way audio
- Dropped calls
- Unregistered phones
- Failed transfers
Network Review Checklist
Below is a quick overview of all recommendations for your network to utilize NEXT without issues. Depending on your equipment, not all settings may be present.
- Bandwidth and Jitter meet recommendations
- SIP ALG is disabled
- No Double NAT on network
- UDP 10000–65535 allowed outbound (for RTP traffic)
- STUN/TURN 443, 3478, 5349 allowed
- SSL inspection bypassed
- FQDNs/IPs allow-listed
- VPN split-tunnel configured
- QoS prioritization (DSCP 46 for RTP) enabled
VPN and Security Inspection
- Use split-tunnel VPN wherever possible.
- If full-tunnel VPN is required:
- Allow outbound UDP media traffic.
- Bypass TLS inspection for STUN/TURN and signaling.
- Avoid proxying or filtering RTP media traffic.
Troubleshooting and Support
Quick Diagnosis
|
Symptom |
Likely Cause |
Quick Fix |
|---|---|---|
| One-way audio | SIP ALG or NAT issue | Disable SIP ALG; allow media ports |
| Dropped calls after 30 seconds | NAT timer expiry | Remove Double NAT |
| Choppy or robotic audio | Jitter or packet loss | Use wired or 5 GHz Wi-Fi; reduce load |
| App rings but does not connect | Content filter or VPN | Switch network or bypass VPN filter |
| Desk phone unregisters | Blocked signaling ports | Verify SIP port access |
WebRTC Diagnostic Tool
If you are experiencing issues with using the NEXT platform, you can run a diagnostic from your machine to identify issues that may exist.
- From the device experiencing the issue, navigate to https://webrtc.nextiva.com/
- Click Start Test if it is your first visit, or Start New Test to run the test again.
- During the Audio Quality section of the test, speak into your microphone; you should hear what you said repeated back over your speakers.
- After the test completes, address any issues identified by the tool.
Troubleshooting Checklist
Start with these basic steps:
- Restart your router or modem (resets NAT tables).
- Switch from Wi-Fi to Ethernet.
- Disable VPN temporarily.
- Move closer to your access point (use 5 GHz).
- Test with a mobile hotspot to isolate LAN issues.
- If the hotspot works, check your local firewall or ISP configuration.
- If issues persist, verify your device and account settings.
Wi-Fi and Device Optimization
- Keep routers away from microwaves, TVs, and solid walls.
- Avoid streaming or large uploads during calls.
- Use wired headsets instead of Bluetooth whenever possible.
Questions from the support team
Before opening a case with support, confirm:
- Is the issue affecting one user or multiple users?
- Is the device wired or wireless?
- Does the issue occur on mobile data?
- Router brand, model, and ISP name?
- Does call quality degrade after transfers?
- Any recent changes to the network or hardware?
FAQs
Do I need to enable inbound port forwarding?
No. Standard outbound access is sufficient for cloud calling.
Why should I disable SIP ALG?
SIP ALG rewrites SIP headers, which often breaks audio or registration.
Can I use public Wi-Fi?
Not recommended. Some public networks block VoIP traffic. Use a cellular hotspot instead if needed.
How much bandwidth is required for 10 concurrent calls?
Plan 3–5 Mbps of dedicated upload and download bandwidth for 10 concurrent voice calls.