Support / Voice / SBC

SBC interop with VoiceTel.

SBC deployments terminate SIP trunks for downstream PBXes, contact-centers, and hosted voice platforms. The interop fields are mostly the same across vendors; the menu paths differ and the codec/SRTP/PRACK behavior varies subtly.

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Common interop pain points

The registration host for your SIP trunk is shown in the customer portal under your trunk's settings. Use UDP transport (most common) or TCP.

  • Authentication mode — VoiceTel supports Digest or IP authentication. Pick one and stick to it; mixed-mode trunks have rare but real edge cases.
  • From username — Always the SIP account username, never the caller ID (see callout above).
  • P-Asserted-Identity — STIR/SHAKEN attestation depends on consistent identity headers. Don't strip P-Asserted-Identity at the SBC.
  • PRACK / 100rel — VoiceTel sends 18x with 100rel for early-media services. Ensure your SBC negotiates PRACK rather than rejecting it.
  • T.38 fax — re-INVITE to T.38 is supported. SBC must not block re-INVITE or strip SDP for fax to land.
  • CallerID format — Use full E.164 in From display-name and P-Asserted-Identity. Local-format CallerIDs may fail STIR/SHAKEN attestation downstream.
  • Codecs and fax — Voice codecs are G.711µ (ulaw), G.711a (alaw), and G.729. Fax is T.38 (re-INVITE during an active G.711 call to switch to T.38/UDPTL).

Vendors

  • AudioCodes Mediant

    Mediant 500/800/1000/2600/4000/9000 hardware and Mediant CE virtual.

  • Cisco CUBE

    Cisco Unified Border Element on ISR, ASR, CSR, or virtual CUBE.

  • FreeSWITCH

    Open-source SBC role via Sofia SIP profile + ACL.

  • Asterisk

    Asterisk PJSIP endpoint as SBC for trunk termination and PBX bridge.