Generic SIP settings
Field names vary between systems. Where your interface uses terms such as SIP server, domain, realm, proxy or registrar, use the values below.
| Setting | Sipflex value |
|---|---|
| User ID / authentication ID | Your Sipflex phone number, for example 442034567890 |
| Display name | Your Sipflex phone number |
| Authentication password | The separate trunk password shown in your portal |
| SIP registrar / server | sip.sipflex.co.uk |
| Outbound proxy | sip.sipflex.co.uk |
| Authentication | Register / account-based |
| DTMF | RFC2833, RFC4733, AVT or telephone-event |
| Registration | Enabled using your Sipflex phone number and trunk password |
Transport and SIP ports
| Transport | Port | Notes |
|---|---|---|
| UDP | 5060 or 50600 | Common default; SIP signalling is unencrypted |
| TCP | 5060 or 50600 | Connection-oriented SIP signalling |
| TLS | 5061 | Encrypted SIP signalling; use TCP port 5061 |
Do not combine TLS with UDP port 5060. If you select TLS, use TCP port 5061 and follow the certificate requirements of your PBX or application.
Media, codecs and DTMF
| Setting | Recommended value |
|---|---|
| Audio codecs | G.711 A-law (PCMA), G.711 µ-law (PCMU), G.722 or G.729a |
| Codec priority | Start with G.711 A-law for standard UK telephony, unless your application requires another supported codec |
| Packet time | 20 ms — 10 ms is not supported |
| DTMF events | RFC2833 / RFC4733 out-of-band |
| Video codecs | H.264 and H.263 where video is deliberately enabled |
Packetisation matters.
Set the audio packet time to 20 ms. A 10 ms packet time is not supported and can cause media problems.
Inbound numbers and outbound caller ID
Create an inbound route for each full number delivered to the connection. For outbound calls, present only a number authorised on your Sipflex account.
Verify the connection
- Confirm the SIP account reports Registered or Online.
- Place an outbound call and verify two-way audio and the caller ID received.
- Call each Sipflex number routed to the connection from another network.
- Test an IVR or keypad flow to confirm RFC2833/RFC4733 DTMF.
- If registration succeeds but audio fails, inspect NAT, RTP and firewall handling on your system.
Support scope
Sipflex can confirm credentials, registration state and provider-side routing. Configuration of your PBX, firewall, dial plan or application remains your responsibility.