In short
A SIP trunk is an external line service that connects a company’s IP PBX to the telephone network over the internet or a private circuit instead of physical lines. The trunk is bought from a carrier authorised in Türkiye. CybUP connects it to PBXs such as 3CX and Asterisk/FreePBX, configures codecs, NAT, the firewall and a backup line, and hands over after call testing.
What is a SIP trunk and how does SIP trunking work?
A SIP trunk is a virtual bundle of external lines over which the carrier talks to your PBX using the SIP protocol. Call setup and teardown travel as SIP (RFC 3261) messages, and the audio itself as RTP (RFC 3550) packets. One connection can carry several calls at once; the number of concurrent channels and the number block are set in your contract with the carrier.
One important distinction: CybUP does not supply the line. The SIP trunk is bought from a carrier authorised in Türkiye to provide fixed telephony. We take the connection details the carrier gives you and make your PBX, network and firewall work with that trunk. If you do not have a PBX yet, IP PBX installation comes first.
“This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.”
What should you ask a carrier before buying a SIP trunk?
At quote stage, ask about the number of concurrent channels, the number block, the authentication method, supported codecs, and whether the connection comes over the internet or over the carrier’s own circuit. The answers decide how the setup is done.
Authentication works in one of two ways: the PBX registers with the carrier using a username and password (registration), or the carrier accepts traffic only from your static IP address (IP-based). 3CX’s SIP trunk documentation covers the two methods separately. IP-based authentication needs a static public IP address.
- Number of concurrent channels and number block (DIDs)
- Authentication: registration or IP-based
- Supported codecs and fax support
- Delivered over the internet or over a carrier circuit
- Caller ID presentation rules
“A SIP trunk authentication is either register based (You will have account details) or IP based, in which case you will need to specify an IP.”
Why do codec, NAT and firewall settings matter so much?
Most SIP trunk faults are not in the PBX but in the network device in between. The classic symptom is one-way audio: you can hear the other party, but they cannot hear you. The usual cause is a PBX behind NAT writing its own internal IP address into the SIP message, and the carrier then trying to send the audio to that address.
To prevent this, the PBX is configured with the correct public IP address, a defined RTP port range is opened on the firewall, and SIP traffic is accepted only from the carrier’s IP addresses. SIP ALG, which is switched on by default on many firewalls, rewrites SIP messages and can cause problems; we turn it off in most installations. On networks with a FortiGate, we handle these settings together with FortiGate installation.
On codecs, most carriers work with G.711: audio quality is high, and so is the bandwidth it needs. Compressed codecs such as G.729 use less bandwidth, but not every carrier supports them. If the shared codec order between the PBX and the carrier is not set correctly, calls fail to connect or get transcoded for no reason.
Do you need an SBC?
An SBC (Session Border Controller) is an appliance or piece of software that sits between the PBX and the carrier and polices SIP traffic. In small and mid-sized installations, a properly configured firewall plus the PBX’s own security settings is usually enough.
Where several PBXs share one trunk, where there are many remote users, or where the carrier requires a particular SIP format, an SBC makes management easier. We decide whether you need one after reviewing the existing network.
How does number porting work in Türkiye?
You can port your existing fixed-line numbers to the new SIP trunk carrier. Under the number portability procedures of BTK, Türkiye’s telecoms regulator, you apply to the operator you want to move to (the recipient operator), and fixed number portability covers both geographic and non-geographic numbers.
On the technical side, porting day is the critical moment. The PBX is prepared in advance with inbound routing rules for the numbers being ported, so the moment the carrier takes over a number, calls land on the right extension or IVR. While the port is going through, it is normal for calls to arrive on the old line for a short while and then on the new one, which is why we recommend scheduling the switchover out of hours.
What happens if the SIP trunk goes down, and how do you set up a backup?
Because a SIP trunk depends on the internet or a carrier circuit, external calls stop when that connection drops. The way to avoid this is redundancy on at least two levels: the network connection and the external line itself.
On the network side, you can add a second internet line and a firewall that fails over automatically; we cover this in detail on the FortiGate SD-WAN and dual WAN page. On the line side, a second trunk or a GSM gateway is defined and the outbound rules are written to move to it when the primary drops. For inbound calls, the forwarding options the carrier offers should also be discussed.
Say a 25-person insurance agency cannot afford to lose customer calls. A SIP trunk over the fibre line is the primary route, with either a trunk from the same carrier over a second internet line or a GSM gateway as backup; when the main line drops, outbound calls go out over the backup path automatically. We work out which option suits you in the free review.
What you receive
- SIP trunk defined on the PBX with the carrier’s details, plus registration or IP authentication
- Inbound (DID) and outbound call routing rules
- Codec order and fax settings
- Firewall/NAT rules: carrier IP restriction, RTP port range, SIP ALG setting
- Automatic failover to a backup trunk or GSM gateway, where needed
- Call test report: internal, external, mobile, international restriction and one-way audio checks
How we work
- 1
Initial review
We review your PBX, firewall, internet connection and the details provided by the carrier.
- 2
Carrier coordination
We confirm authentication, codec and IP details with the carrier and, where needed, ask the technical questions ourselves.
- 3
Configuration
We define the trunk on the PBX, set up the firewall and NAT, and write the call routing rules.
- 4
Testing
We test inbound, outbound, mobile and long-duration calls, and check audio quality and failover.
- 5
Go-live
We bring the numbers live at the scheduled time and watch the call logs on the first day.
Frequently asked questions
Does CybUP supply the SIP trunk line?
No. The SIP trunk is bought from a carrier authorised in Türkiye, and the contract is between you and the carrier. We connect that trunk to your PBX and network and configure everything it needs.
How long does SIP trunk setup take?
With the carrier’s details in hand, configuration and testing are usually finished within one working day. If numbers are being ported, the carrier’s process sets the overall timeline.
We have one-way audio on our calls. Can you fix it?
Yes. It is the SIP trunk problem we see most often, and it usually comes down to NAT, SIP ALG or RTP port settings. We review the PBX and firewall settings, find the cause and fix it.
Which PBXs do you work with?
We configure trunks on SIP-capable IP PBXs, chiefly 3CX and Asterisk/FreePBX. If you run a different PBX, we look at its model and version and assess it.
Will there be an outage if we port our existing number?
There can be a short transition at the moment of porting. Because we prepare the PBX in advance and schedule the switchover out of hours, the impact is usually limited.
Do we need a static IP address for a SIP trunk?
Yes, with carriers that use IP-based authentication. With carriers that use registration it may not be mandatory, but we still recommend one because it makes the security rules easier to write.
How much does SIP trunk setup cost?
We give a written quote once we have reviewed the scope; the review is free. Line charges are set directly by the carrier under its own tariff.
Can SIP trunk setup be done remotely?
Yes. With remote access to the PBX and firewall, SIP trunk setup can be done remotely anywhere in Türkiye. In Istanbul we come on site when needed.