What a SIP trunk does
A SIP trunk connects your phone system to the public telephone network over the internet. Your PBX, contact-centre dialer or softswitch sends and receives calls using the Session Initiation Protocol for signalling and the Real-time Transport Protocol, RTP, for the audio itself. Telnxo terminates your outbound calls to the carrier network and delivers inbound calls to your equipment.
Setting up a trunk is mostly about agreeing on four things with the provider: how each side proves who it is, where signalling goes, which audio formats are used and which numbers may appear as caller ID. Get those four right and calls just work. Get one wrong and you will see the classic symptoms: calls that ring and then drop, calls with no audio and calls that are rejected before they start.
What you need before you start
- The public IP address of your PBX or session border controller
- The SIP host name and port shown in your Telnxo dashboard
- Authentication details: either your IP allow-listed or a username and password
- The numbers or caller IDs you are permitted to present
- Firewall rules for SIP signalling and the RTP media range
- A way to capture a call trace: your PBX logs or a packet capture tool
Connect step by step
- Choose the authentication method. IP allow-listing is simplest and most robust when your PBX has a fixed public address. Username and password registration suits equipment that changes address. Do not mix both on one trunk without a reason
- Add the outbound trunk. In your PBX create a trunk pointing at the Telnxo host and port, using the transport you agreed: UDP, TCP or TLS
- Add the inbound peer. Create a matching peer or endpoint so that calls arriving from Telnxo are accepted and routed to the right extension or queue
- Set codecs. Offer G.711 first for quality. Add others only as needed
- Set the caller ID rules. Decide which number is presented on outbound calls and on which routes
- Place test calls. Call out to a mobile phone, then call in from one. Confirm audio in both directions for at least thirty seconds
- Raise capacity. Set your channel limit above your peak concurrent calls, with headroom
Choosing codecs
A codec turns your voice into data and back. The choice trades quality against bandwidth and compatibility.
| Codec | Quality | Bandwidth per call | Notes |
|---|---|---|---|
| G.711 (PCMU or PCMA) | Good, telephone quality | About 87 kbps with overhead | Universally supported; the safe default |
| G.729 | Acceptable | About 31 kbps with overhead | Saves bandwidth; check licensing on your equipment |
| Opus | Very good | Variable | Used mainly where both ends support it |
Always offer the same codecs on both sides, in the same order of preference. A mismatch is a frequent cause of calls that connect and then carry silence.
Caller ID
Present only numbers you own or are authorised to use. Calls with unverified or malformed caller IDs may be blocked, labelled as spam or rejected by the destination network. Use the E.164 format, which is a plus sign, the country code and the national number with no spaces, for example +911412345678.
- Outbound caller ID should match a number you have registered with Telnxo for that account.
- Inbound numbers are delivered to you in the format described in your dashboard; normalise them in your PBX.
- Regulation in India and other regions restricts how business calls may be presented, so confirm local rules before you choose.
Firewall and network
SIP signalling uses a fixed port, usually 5060 for UDP and TCP and 5061 for TLS. RTP uses a range of high UDP ports negotiated per call. Open the signalling port to the Telnxo addresses and the media range in both directions. Disable any SIP ALG or similar feature on your router: it rewrites packets in ways that often break calls.
Troubleshooting by symptom
| Symptom | Likely cause | What to check |
|---|---|---|
| Call rejected immediately | Authentication or number format | IP allow-list, credentials, dialled number in E.164 |
| Rings, then drops | Signalling or timers | SIP trace; look for the response code |
| Connected, no audio | Media blocked or NAT | RTP port range, SIP ALG, public IP in SDP |
| One-way audio | NAT or firewall in one direction | Open media in both directions; use the public address |
| Robotic or choppy audio | Packet loss or jitter | Network quality, codec choice, QoS marking |
| Echo | Poor acoustics or double processing | Disable duplicate echo cancellation |
Capacity and monitoring
Each concurrent call uses one channel. Estimate peak concurrency from call volume and average duration: calls per hour multiplied by average minutes, divided by sixty. Add twenty per cent headroom. Watch three numbers every day: concurrent calls against your limit, answer-seizure ratio and average call duration. A sudden change in any of them is the first sign of a routing or fraud problem.
Security
- Allow-list Telnxo addresses on your firewall and nothing wider
- Use strong passwords if you register with credentials
- Set a spending and channel limit to cap damage from toll fraud
- Block international destinations you never call
- Keep PBX software patched
Reading a SIP trace
When a call fails, a trace tells you exactly why. You do not need to read every line. Look for the first response to the INVITE that starts the call.
| Response | Meaning | Typical fix |
|---|---|---|
| 100 Trying | Request received | None; wait for the next response |
| 180 or 183 | Ringing or early media | Normal progress |
| 200 OK | Call answered | Check audio next |
| 401 or 407 | Authentication required | Credentials or IP allow-list |
| 403 Forbidden | Not permitted | Caller ID, destination or account limit |
| 404 Not Found | Number not routable | Dialled format or prefix |
| 480 or 503 | Temporarily unavailable or overloaded | Retry; check channel limit |
Capture the trace on your side as well as asking support to read ours. Comparing the two shows where a message was changed or lost on the way.
Go-live plan
- Test in a quiet window. Place twenty test calls in each direction at a time when a failure will not affect customers
- Move a small share of traffic. Route about ten per cent of calls through the trunk for a day
- Watch the numbers. Answer-seizure ratio, post-dial delay and audio complaints
- Increase gradually. Move to fifty per cent, then everything, when numbers are stable
- Keep a fallback. Leave your previous route configured for a week in case you need to switch back
Further reading
- RFC 3261, SIP: Session Initiation Protocol
- RFC 3550, RTP: A Transport Protocol for Real-Time Applications
- ITU-T E.164, the international numbering plan
Frequently asked questions
UDP or TCP?
Either works. UDP is common; use TCP or TLS if your network drops large UDP packets or you need encryption.
Why do calls ring but fail to connect?
Check authentication, the dialled number format and whether the caller ID is allowed on the trunk.
How many channels do I need?
Work out peak concurrent calls and add headroom. Each active call uses one channel.
Can I use my own numbers?
Presenting a number you own is possible subject to ownership checks and local rules. Contact support to have it verified.
Do I need a session border controller?
Not always, but one helps with security, NAT handling and codec transcoding at larger scale.