SIP Interconnection With Wise Network
A SIP interconnection connects your softswitch or session border controller directly to ours, so the two networks can exchange voice traffic in both directions.
What is SIP interconnection?
SIP interconnection is a direct, IP-based link between two carriers' voice switches that uses the Session Initiation Protocol (SIP) to set up calls and RTP to carry the audio. It replaces the older TDM interconnect. Once the two sides have exchanged IP addresses, agreed codecs and tested calls, voice traffic can flow between the networks for termination, origination or both.
Technical requirements
| Item | What we exchange |
|---|---|
| Signalling | SIP over UDP or TCP, with the signalling IP addresses and ports of both sides |
| Media | RTP media IP ranges, so firewalls on both sides allow audio |
| Codecs | Agreed list, normally including G.711; others by agreement |
| Number format | E.164 international format, with any technical prefix agreed in advance |
| Capacity | Concurrent call and calls-per-second limits agreed for the expected traffic |
| Authentication | IP-based access lists on both sides |
Onboarding steps
- Agreement and forms. Commercial agreement, company documents and the technical interconnection form.
- Exchange of IP details. Both sides open their firewalls and SBC access lists to the other's signalling and media addresses.
- Configuration. Trunks, codecs, number format and capacity limits are configured on both switches.
- Test calls. Calls to agreed test numbers confirm routing, two-way audio, caller ID and billing records.
- Go-live. Rates are loaded and live traffic starts, usually with monitored volumes at first.
Security on SIP trunks
SIP trunks are a common target for toll fraud and scanning. We accept traffic only from addresses agreed during onboarding, apply capacity limits per partner, and watch for unusual destination patterns and short-duration call bursts. We recommend the same controls on your side.
After go-live
Traffic, CDRs and statements are visible through our web interface. Route quality is monitored per destination, and our technical support team, available 24/7, handles faults and changes through agreed contacts.
Who it is for
Voice engineers at carriers, ITSPs and aggregators who are connecting to Wise Network for termination, origination or both.
Frequently asked questions
Do you authenticate SIP traffic by IP or by credentials?
Carrier interconnects are normally authenticated by IP address on both sides, with traffic accepted only from the addresses exchanged during onboarding. The method is confirmed in the interconnection form.
Which codecs do you support?
G.711 is the common baseline for wholesale interconnects; compressed codecs such as G.729 are agreed per partner. We confirm the codec list during onboarding.
How long does an interconnection take?
It depends mostly on paperwork and testing on both sides. The technical set-up is short once IP addresses and codecs are exchanged; test calls then confirm routing, audio and caller ID before live traffic.
Can we use the same interconnection for origination and termination?
Yes. One SIP interconnection can carry traffic in both directions, each with its own rate plan.
Related
Start a SIP interconnection
Send us your company details and the traffic you plan to exchange. Our engineers will send the interconnection form and our signalling and media details.
- Email: sales@wisenetwork.co
- Phone: +961 3 085 999