SIP Server Settings
Use these settings to configure your trunk or SIP device:
| Setting | Value |
|---|---|
| SIP Server / Registrar | yourdomain.s1.ipcomms.net |
| Outbound Proxy | yourdomain.s1.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.
Codec Configuration
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 Settings
| DTMF Mode: | RFC 2833 (recommended) |
| Payload Type: | 101 |
Firewall Configuration
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.
NAT Configuration
If your PBX is behind a NAT router, configure these settings:
Option 1: External IP
Set your external (public) IP address in your PBX settings.
Option 2: STUN Server
Use a STUN server for dynamic NAT detection.
Common NAT Issues
One-Way Audio
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.
Dropped Calls After 30 Seconds
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.
Registration Failures
Trunk shows as "Unavailable" or fails to register.
Solution: Verify credentials, check port 5060 is open, confirm DNS resolution works.
Outbound Caller ID
Configure your outbound caller ID in your PBX:
Format Requirements
- Caller ID Number: Use one of your IPComms DIDs in E.164 format (e.g., +14045551234)
- Caller ID Name: Set via CNAM in your portal (15 characters max)
- P-Asserted-Identity: Optionally include for privacy-aware routing
Note: You can only use phone numbers assigned to your account. Attempts to spoof other numbers will be blocked.
Platform-Specific Guides
Detailed configuration guides for popular PBX platforms:
Testing Your Configuration
1. Verify Registration
Check that your trunk shows as "Registered" in your PBX. You can also verify in the IPComms portal.
2. Test Outbound Calls
Make a call to your mobile phone. Verify you can hear audio in both directions.
3. Test Inbound Calls
Call your IPComms number from an external phone. Verify the call routes correctly.
4. Check Caller ID
Verify your outbound caller ID displays correctly on the receiving phone.
Need Configuration Help?
Our support team can assist with your SIP trunk configuration.
(678) 460-1475 support@ipcomms.net