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.
