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SIPTrunkHub

Network

The Network Behind Every Call

Your call reaches our SIP edge, is checked against your account, then routed to a carrier picked for that prefix. If that carrier rejects the call, the next one in the plan gets it. This page explains each step: the protocols, codecs, security and what you see in your CDRs.

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Call path

What happens to a call, step by step

  1. 01

    Edge

    Your INVITE reaches our session border controller. Source IP or digest credentials identify the account.
  2. 02

    Checks

    Balance, channel limit, destination permissions and spend caps are checked before the call goes further.
  3. 03

    Routing

    Longest-prefix match picks the route plan. Your techprefix chooses the tier: premium CLI, standard CLI or non-CLI.
  4. 04

    Failover

    If the first carrier returns a failure, the call is retried on the next carrier in the plan.
Simplified SIP ladder: failover on a 503
Your PBX            SIPTrunkHub SBC         Carrier A          Carrier B
   |--- INVITE ------------>|                       |                  |
   |<-- 100 Trying ---------|                       |                  |
   |                        |--- INVITE ----------->|                  |
   |                        |<-- 503 Unavailable ---|                  |
   |                        |--- ACK -------------->|                  |
   |                        |--- INVITE ------------------------------>|
   |                        |<-- 183 Session Progress -----------------|
   |<-- 183 (early media) --|                       |                  |
   |                        |<-- 200 OK -------------------------------|
   |<-- 200 OK -------------|                       |                  |
   |--- ACK --------------->|--- ACK ---------------------------------->|
   |======= RTP media ======|=========================================>|

Codecs

Codecs and bandwidth per call

Figures are per direction, per call, including IP/UDP/RTP headers at 20 ms packets. Add about 10% for Ethernet framing.

Supported codecs with bandwidth and use
CodecPayload rateBandwidth on the wireUse it for
G.711 a-law / u-law64 kbps~87 kbpsDefault. Best quality, fax pass-through, no transcoding
G.7298 kbps~31 kbpsLimited uplinks, high channel counts. Not for fax
G.72264 kbps~87 kbpsWideband (HD) audio where both ends support it
Opus6–510 kbps (variable)VariesWideband on supported routes; handles packet loss well

Deep dive: G.711 vs G.729 and how many SIP channels you need.

Capabilities

What the network gives you

  • Multi-carrier failover

    Each prefix has a ranked list of carriers. A rejected call moves to the next one inside the same attempt.

  • TLS and SRTP

    Encrypt signalling on 5061 and media with SRTP. Useful when your traffic crosses networks you do not control.

  • IP or digest auth

    Whitelist static IPs, or register with credentials from dynamic addresses. You can use both on one account.

  • T.38 fax

    Fax relay on supported routes, with G.711 pass-through when T.38 is not negotiated.

  • ASR / ACD reporting

    Per-destination answer-seizure ratio and average call duration, so you can judge routes on data.

  • Real-time CDRs

    Each call record appears as the call ends: destination, duration, SIP result code and cost.

Interconnect

Connecting your switch

Most customers connect a PBX or softswitch over the public internet with IP authentication. Carriers and larger wholesale partners can request a dedicated interconnect profile with their own signalling IPs, capacity and codec rules.

Before going live, send an OPTIONS ping to confirm reachability, make a short test call per destination, and check that RTP flows in both directions. Setup steps for each PBX are in the setup guides.

Trunk values (replace with the values from your account)
host        = sip.example-host.net
port        = 5060          ; 5061 for TLS
transport   = udp           ; udp | tcp | tls
auth        = ip            ; or digest
your_ip     = 203.0.113.10  ; whitelisted source
codecs      = alaw,ulaw,g729
dtmf        = rfc4733
techprefix  = 101#          ; selects route tier

FAQ

Network questions

Which codec should I send first?

G.711 (a-law in most of the world, u-law in North America and Japan) gives the best quality and the fewest transcoding issues. Put G.729 second if your uplink is limited. Order matters: we answer with the first codec in your offer that the route supports.

Do you transcode?

Only when the far end cannot accept your codec. Transcoding adds a little latency and can lower quality, so the cleanest result is offering G.711 and G.729 together and letting negotiation pick.

Is fax supported?

Yes, T.38 on routes that support it, with G.711 pass-through as a fallback. Fax over compressed codecs such as G.729 will fail, so disable G.729 on fax endpoints.

What happens if a carrier route fails?

Calls that get a failure response from one carrier are retried on the next carrier in the route plan for that prefix, within the same call attempt. You do not need to resend the call.

Can I connect over TLS?

Yes. Signalling can run over TLS on port 5061 and media over SRTP. Ask the NOC to enable it on your account before switching your trunk.

Want to test the network?

Top up, point a trunk at us and make test calls to your real destinations.
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