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SIPTrunkHub

SIP Trunking

SIP Trunking for PBXs, Call Centres and Dialers

Point your PBX at one SIP endpoint and terminate calls to 50+ countries. Authenticate by IP or credentials, encrypt with TLS and SRTP, and pay per minute from a prepaid balance. No PRI hardware, no contract.

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What a trunk includes

The parts that matter on a live trunk

Everything below is part of the standard account. No feature tiers.

  • Channels on demand

    No fixed bundle of 23 or 30 B-channels. Concurrent call limits are set per account and raised when your traffic grows.

  • Inbound and outbound

    Terminate outbound traffic and receive DIDs on the same trunk, with one balance and one set of CDRs.

  • IP or digest auth

    Whitelist your public signalling IPs, register with username and password, or combine both for remote sites.

  • TLS and SRTP

    Encrypt signalling on TCP/5061 with TLS and media with SRTP when your PBX supports it. Plain UDP/5060 remains available.

  • Multi-carrier failover

    Each destination is routed across more than one upstream carrier. A failed or degraded leg is skipped on the next attempt.

  • Real-time CDRs

    Every call record lands in the portal as it ends: duration, disposition, SIP cause, cost. ASR and ACD per destination.

Call flow

What happens on an outbound call

Your PBX sends an INVITE with its codec offer. Our SBC authenticates it, picks the route for the dialled prefix and forwards the INVITE upstream. If the first carrier rejects the call with a 5xx or times out, the SBC retries on the next carrier before your caller hears anything.

Media flows between your PBX and the SBC, so you only whitelist our addresses. Read more in What Is a SIP Trunk?

SIP ladder · outbound call
PBX                     SIPTrunkHub SBC                Carrier
 |  INVITE (SDP offer)    |                              |
 |----------------------->|                              |
 |  100 Trying            |  INVITE (SDP offer)          |
 |<-----------------------|----------------------------->|
 |                        |  183 Session Progress        |
 |  183 (early media)     |<-----------------------------|
 |<-----------------------|  200 OK (SDP answer)         |
 |  200 OK                |<-----------------------------|
 |<-----------------------|                              |
 |  ACK                   |  ACK                         |
 |----------------------->|----------------------------->|
 |<=========== RTP / SRTP media (G.711 / G.729) ========>|
 |  BYE                   |  BYE                         |
 |----------------------->|----------------------------->|

Authentication

IP authentication or digest registration

Pick per site. Many customers use IP auth for the data-centre PBX and digest for branch offices.

IP authentication compared with digest authentication
IP authenticationDigest / registration
Best forStatic public IP, data-centre PBX, dialersDynamic IP, small offices, softphones
How it worksWe whitelist your signalling IPsPBX registers with username and password
Tech prefixUsed to select route tierOptional, per account
Toll-fraud exposureLow: no password to leakUse strong passwords and IP restriction
NATUsually not an issueKeep-alives required (≤ 60 s)

Moving from PRI / ISDN

A migration that does not drop calls

Run the SIP trunk in parallel, move routes one by one, then retire the PRI. See SIP trunk vs PRI.

  1. 01

    Audit

    List current PRI channels, peak concurrent calls, numbers in use and fax lines.
  2. 02

    Connect in parallel

    Add the SIP trunk to the PBX next to the PRI. Test outbound and a test DID inbound.
  3. 03

    Move outbound first

    Change outbound routes to SIP by destination. Keep PRI as the failover route.
  4. 04

    Move numbers, retire PRI

    Port or replace inbound numbers, watch CDRs for a full billing cycle, then cancel the PRI.
pjsip.conf · placeholder values
; Replace host and IP with the values from your account.
[siptrunkhub]
type=endpoint
transport=transport-udp
context=from-siptrunkhub
disallow=all
allow=ulaw,alaw,g729
aors=siptrunkhub
direct_media=no
rtp_symmetric=yes
force_rport=yes

[siptrunkhub]
type=aor
contact=sip:sip.example-host.net:5060
qualify_frequency=60

[siptrunkhub]
type=identify
endpoint=siptrunkhub
match=sip.example-host.net

Setup guides

Config for the PBX you already run

Step-by-step guides with real config, codec order, NAT notes and an error table.

FAQ

SIP trunking questions

How many channels do I get?

Channels are not sold in bundles. Your account has a concurrent call limit that starts at a level matched to your stated traffic and is raised on request once traffic is stable.

Can I keep my existing numbers when moving from PRI?

Porting depends on the country and the losing carrier. Send sales the numbers and country and we confirm whether they can be ported or whether new DIDs are the faster route.

Do you support registration for PBXs behind dynamic IPs?

Yes. Digest authentication with registration works behind dynamic IPs. Set the PBX to keep the NAT binding alive with OPTIONS or re-REGISTER every 60 seconds or less.

Which transport and ports should I open?

Signalling on UDP or TCP 5060, or TLS on 5061. RTP uses a UDP range that is listed in your account. Disable SIP ALG on the router in front of your PBX.

Is fax supported?

T.38 is supported with G.711 passthrough as a fallback. Fax over IP is sensitive to jitter and packet loss, so test the specific destinations you need before relying on them.

Is there a contract or setup fee?

No. The account is prepaid: top up at least $200 and the balance is used per call. You can stop sending traffic at any time.

Tell us your destinations and channel count

Sales replies with routes, rates and a test window. Prepaid from $200.
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