Configure your PBX or softphone to connect to IPComms SIP Trunking services.
Use these settings to configure your trunk or SIP device:
| Setting | Value |
|---|---|
| SIP Server / Registrar | sip.ipcomms.net |
| Outbound Proxy | sip.ipcomms.net |
| SIP Port (UDP/TCP) | 5060 |
| SIP Port (TLS) | 5061 |
| Transport | UDP (recommended), TCP, or TLS |
| Username | Your SIP username (from portal) |
| Password | Your SIP password (from portal) |
| Auth Username | Same as Username |
| Registration | Required (60-300 seconds) |
Note: Your specific SIP credentials are available in the customer portal under SIP Trunking → Credentials.
Configure codecs in priority order for best quality:
| Priority | Codec | Bandwidth | Notes |
|---|---|---|---|
| 1 | G.711u (PCMU) | 87 Kbps | Best quality, recommended |
| 2 | G.711a (PCMA) | 87 Kbps | Equivalent quality, Europe standard |
| 3 | G.729 | 31 Kbps | Low bandwidth, good for limited connections |
| DTMF Mode: | RFC 2833 (recommended) |
| Payload Type: | 101 |
Open the following ports in your firewall for SIP and RTP traffic:
| Purpose | Protocol | Port(s) | Direction |
|---|---|---|---|
| SIP Signaling | UDP | 5060 | Inbound & Outbound |
| SIP Signaling (TLS) | TCP | 5061 | Inbound & Outbound |
| RTP Media | UDP | 10000-20000 | Inbound & Outbound |
Important: If you're behind NAT, ensure your PBX is configured to use STUN or has proper NAT traversal settings to avoid one-way audio issues.
If your PBX is behind a NAT router, configure these settings:
Set your external (public) IP address in your PBX settings.
external_media_address = YOUR_PUBLIC_IP
external_signaling_address = YOUR_PUBLIC_IP
Use a STUN server for dynamic NAT detection.
stun_server = stun.l.google.com:19302
You can hear the caller, but they can't hear you (or vice versa).
Solution: Check NAT settings, verify RTP ports are open, ensure external IP is configured.
Calls connect but disconnect after about 30 seconds.
Solution: Ensure SIP packets can return through your firewall. Enable SIP ALG or use a SIP-aware firewall.
Trunk shows as "Unavailable" or fails to register.
Solution: Verify credentials, check port 5060 is open, confirm DNS resolution works.
Configure your outbound caller ID in your PBX:
Note: You can only use phone numbers assigned to your account. Attempts to spoof other numbers will be blocked.
Detailed configuration guides for popular PBX platforms:
Check that your trunk shows as "Registered" in your PBX. You can also verify in the IPComms portal.
Make a call to your mobile phone. Verify you can hear audio in both directions.
Call your IPComms number from an external phone. Verify the call routes correctly.
Verify your outbound caller ID displays correctly on the receiving phone.
Our support team can assist with your SIP trunk configuration.