Three real-world scenarios that VoIP engineers see every week — analyzed by SIPSymposium with severity-ranked findings and concrete recommendations. No signup needed to view.
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14:23:51.318 the carrier sent 200 OK accepting the INVITE. No ACK was received within the Timer F window of 32 seconds (T1×64 with T1=500ms). The carrier retransmitted 200 OK 6 times using exponential backoff per RFC 3261. At 14:24:23.812 the carrier terminated the dialog with a BYE carrying Reason: SIP;cause=408 — this 32.5-second gap is the diagnostic fingerprint of ACK timeout, not a network or carrier-side failure.carrier-edge-3.example.net. Per RFC 3261 the PBX must send ACK to that Record-Routed target, not the original carrier address. The absence of any ACK in the trace combined with the carrier's continued retransmission indicates the ACK is either:
Contact: <sip:10.0.4.21:5060> exposes a private LAN address to the carrier. Even after the ACK routing is fixed, this Contact prevents in-dialog requests from the carrier (re-INVITE, UPDATE, BYE) from reaching the PBX directly — they will rely entirely on the Record-Route path. Recommend an SBC or border element that rewrites Contact to the public-facing address.192.168.1.15 across the public internet. The PBX is not aware of its public-facing NAT address.203.0.113.42:10000 shows 1,500 packets over 30 seconds, matching the expected G.711µ rate of 50pps. Inbound RTP at the PBX listener (192.168.1.15:16384) shows zero packets. This is unambiguously one-way audio in the carrier→PBX direction.c= line advertises private (RFC 1918) IP as the media endpointc=IN IP4 192.168.1.15 specifies a private LAN address as the carrier's destination for RTP. The carrier cannot route packets to this address across the public internet — they will be discarded at the first public router. Root cause: the PBX is not configured with its public NAT address and is using its private interface IP in SDP.externaddr / externhost; FreeSWITCH: ext-rtp-ip / ext-sip-ip; FreePBX: NAT settings under SIP Settings.m=audio line lists payload types 18 8 0 — G.729 first, then G.711A and G.711µ. Microsoft Teams Direct Routing accepts only SILK and G.711 (A-law / µ-law) on the SIP trunk leg. G.729 is explicitly not supported, and Teams will reject the entire offer with 488 when G.729 is in the codec list and the SBC is not transcoding upstream. The 488 response includes Warning: 305 "Incompatible codec", which is Microsoft's specific signal for codec rejection.annexb=no on G.729 — silence suppression disabledfmtp:18 annexb=no parameter disables G.729 Annex B (VAD/CN). This is unusual for Teams interop configurations and may indicate the SBC's codec profile is configured for a different downstream environment that was carried over without adjustment.learn.microsoft.com/microsoftteams/direct-routing-border-controllers.SIPSymposium accepts raw SIP, PCAP files, or natural-language scenario descriptions and returns the same analysis depth shown above.