Supported SIP devices
Most standard SIP-compatible desk phones work with the platform, including:- Polycom VVX series
- Yealink T series
- Cisco SPA series
- Grandstream GXP series
- Fanvil X series
Any device that supports standard SIP registration should work. If your specific device is not listed, try configuring it with the SIP credentials — most modern VoIP phones are compatible.
Configuration steps
Get your SIP credentials
Go to Settings > Phone System > SIP and copy your SIP credentials:
- SIP Server / Registrar address
- SIP Username
- SIP Password
- Port number
Access your phone's admin panel
Connect your desk phone to your network. Open the phone’s web admin interface by entering its IP address in your browser. Refer to your phone manufacturer’s documentation for the default IP and login credentials.
Enter SIP settings
In the phone’s SIP account settings, enter:
- SIP Server/Registrar: the server address from the platform
- SIP User ID / Auth ID: your SIP username
- Password: your SIP password
- Outbound Proxy: same as the SIP server (unless specified otherwise)
- Transport: UDP (or TCP if required)
Set audio codecs
Configure audio codecs in this priority order:
- G.711u (PCMU) — best quality for local calls
- G.711a (PCMA) — alternative for international
- G.729 — lower bandwidth, acceptable quality
Firewall requirements
Ensure your network firewall allows traffic on these ports:| Port | Protocol | Purpose |
|---|---|---|
| 5060 | UDP/TCP | SIP signaling |
| 5061 | TLS | Secure SIP signaling |
| 10000-20000 | UDP | RTP media (audio) |
Troubleshooting no audio
One-way audio
One-way audio
Usually caused by NAT (Network Address Translation) issues. Enable STUN on the phone or configure your router’s SIP ALG (Application Layer Gateway). Some routers require SIP ALG to be disabled — try toggling it.
No audio in either direction
No audio in either direction
Verify that UDP ports 10000-20000 are open on your firewall. If using a corporate network, contact your IT team to open RTP media ports.
Phone registers but calls fail
Phone registers but calls fail
Check that the SIP transport protocol matches (UDP vs TCP). Also verify the outbound proxy setting is correct.
Choppy or robotic audio
Choppy or robotic audio
Typically a bandwidth issue. Ensure your network has sufficient upload bandwidth (at least 100 kbps per concurrent call). Use G.729 codec for lower bandwidth environments. Prioritize VoIP traffic with QoS settings on your router.
Recommended devices
| Device | Price range | Best for |
|---|---|---|
| Yealink T33G | Budget | Small offices and remote workers |
| Polycom VVX 250 | Mid-range | Professional offices |
| Yealink T54W | Mid-range | Power users (color display, Wi-Fi, Bluetooth) |
| Grandstream GXP2170 | Mid-range | Call centers (12 line keys, dual Gigabit ports) |
Frequently asked questions
Can I use a softphone app instead of a physical desk phone?
Can I use a softphone app instead of a physical desk phone?
Yes. Any SIP softphone app (Zoiper, Linphone, Bria) can connect using the same SIP credentials. Configure the app the same way you would a physical phone.
How many desk phones can I connect?
How many desk phones can I connect?
Each SIP user/extension can be registered on one device at a time. To connect multiple desk phones, set up additional SIP users for each phone.
Can I use the desk phone and web dialer at the same time?
Can I use the desk phone and web dialer at the same time?
The web dialer and SIP desk phone operate on separate channels. Incoming calls can be configured to ring one or both depending on your call routing settings.