A SIP interconnect, also called SIP trunk peering, is a trunk between two carriers' switches: the buyer sends calls from authorized IP addresses to the seller's SIP address, both sides agree on codecs, number format and limits, and a few test calls confirm that signaling, audio and CLI work. Most interconnects are live within a day or two once the details are exchanged.
The details both sides exchange
| Item | Example | Why it matters |
|---|---|---|
| Signaling IP addresses | Your SBC or switch public IPs | The seller allows calls only from these (IP authentication) |
| Media IP addresses | Where RTP audio comes from | Firewalls must allow audio, not just signaling |
| SIP address and port | Seller's IP, port 5060 (UDP or TCP) or 5061 (TLS) | Where to send the INVITE |
| Tech prefix | A few digits added before the number | Identifies your account or route on shared IPs |
| Number format | E.164 without plus, such as 447700900123 | Wrong format means failed calls |
| Codecs | G.711 A-law, G.711 mu-law, G.729 | Avoids transcoding and audio failures |
| DTMF method | RFC 2833 / 4733 | Key presses must reach IVRs |
| Capacity | Channels and CPS | Calls above the limit are rejected |
Authentication
Carriers almost always authenticate wholesale trunks by source IP address. The seller adds your IPs to an access list, and calls from anywhere else are refused. Some also support username and password (SIP digest) registration, but IP authentication is standard between carriers. Keep the list small and tell your supplier at once if an IP changes. Weak trunk security is a common entry point for toll fraud.
Number format and dial plan
Agree exactly how numbers are sent: with or without a plus sign, with a tech prefix, and how the CLI is formatted. The standard is E.164, and normalizing your dial plan to it before the call leaves your switch avoids most routing errors.
Codecs, timers and SIP options
- Offer the codecs you support in order of preference. See G.711 vs G.729.
- Agree on session timers or keepalives, so long calls are not cut by a timeout on one side.
- Agree on SIP OPTIONS pings if either side uses them to monitor the trunk.
- Decide how caller ID is carried (From header, P-Asserted-Identity) and whether privacy headers are honored.
Testing before live traffic
- Place a test call to a number you control in the destination country. Check that it rings, that audio goes both ways and that the CLI arrives as expected.
- Test several operators and both mobile and fixed numbers.
- Check that your switch and the seller's CDRs match on duration for the same calls.
- Start with a small share of traffic and watch ASR, ACD and PDD before increasing.
The guide on testing a VoIP route goes into detail, and one-way audio troubleshooting covers the most common setup problem: NAT or firewall rules that block the media stream.
Paperwork around the trunk
Before the technical setup, carriers usually sign an interconnect agreement and complete KYC checks. See the interconnect agreement checklist.
Getting connected
With VoIPMatcher you set up the interconnect once and we manage the route behind it with our carriers. Request a voice route or read more about our VoIP termination service.
Frequently asked questions
What is the difference between a peer SIP trunk and a register SIP trunk?
A peer trunk is authenticated by source IP address: the provider accepts calls only from IPs on its list, and no registration is needed. A register trunk logs in with a username and password and keeps that registration alive. Carriers almost always peer by IP; registration is more common for PBXs on dynamic addresses.
What is SIP peering?
SIP peering is a direct SIP connection between two networks that exchange calls, without registration and usually authenticated by IP address. Between carriers it is the normal way to buy and sell wholesale routes. The two sides agree on IPs, ports, codecs, number format, capacity and caller ID handling, then test before live traffic.
Which port does SIP use?
SIP signalling normally uses port 5060 over UDP or TCP and port 5061 over TLS. Audio travels separately as RTP on a range of UDP ports set on each side's equipment. Carriers state their signalling and media addresses and ports during setup, and your firewall must allow both from their IPs.
Is UDP or TCP better for a SIP trunk?
UDP is the most common transport between carriers and works well on clean links. TCP helps when SIP messages are large enough to be fragmented, and TLS on port 5061 encrypts the signalling. Use what both sides support and record it in the interconnect details, so retransmission and timeout behaviour match on both ends.
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