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VoIP protocol guide
Signalling and media are different
Most IP voice systems use one protocol family to negotiate a call and another to carry the audio. A call can therefore have successful signalling but no usable media.
| Protocol | Main job | What to inspect |
|---|---|---|
| SIP | Creates, modifies, and ends sessions | Registration, INVITE, responses, SDP, routing |
| SDP | Describes media parameters inside signalling | IP/port, codecs, direction, packetization |
| RTP | Carries real-time audio/video | Sequence numbers, timestamps, payload type |
| RTCP | Reports media statistics and synchronises streams | Loss, jitter, round-trip information |
| SRTP | Encrypts and authenticates RTP | Keying, cipher profile, replay protection |
| TLS | Protects signalling transport | Certificate trust, names, versions, expiry |
| STUN | Helps discover public-facing address/port mappings | NAT binding and reflexive candidates |
| TURN | Relays media when direct traversal fails | Relay allocation and bandwidth |
| ICE | Tests candidate paths and selects a working route | Candidate pairs and connectivity checks |
SIP transport basics
- UDP is lightweight but has no transport-level delivery guarantee.
- TCP provides an ordered stream and can help with larger messages or restricted networks.
- TLS protects SIP signalling in transit; it does not automatically encrypt RTP media.
- A SIP
200 OKmeans a transaction succeeded, not that the user heard audio.
The SDP clue
When diagnosing one-way audio, compare the media address and ports advertised in SDP with the address and ports actually reachable from the other side. NAT, stale contact information, symmetric RTP assumptions, and firewalls commonly break this boundary.