Most VoIP troubleshooting becomes easier when signaling, media, audio encoding, and network capacity are treated as separate layers.
SIP controls the conversation
SIP commonly registers endpoints and manages call setup, ringing, answer, transfer, and hang-up. A successful SIP exchange proves that the parties negotiated a session; it does not prove that audio can travel.
RTP carries the voice
After the call is established, RTP normally carries small packets of encoded audio. RTCP may report timing, packet loss, and jitter. Firewalls must allow the negotiated media path, and the addresses announced in SDP must be reachable.
Codecs trade bandwidth for processing and quality
G.711 uses more bandwidth and is widely associated with clear narrowband telephone audio. Compressed codecs reduce bandwidth but may require more processing and can be less tolerant of repeated transcoding. G.722 and other wideband codecs can provide richer audio when both sides support them.
A codec mismatch may prevent the call or force a gateway to transcode. Every conversion adds processing and can reduce quality, so a simple end-to-end codec path is preferable.
Plan bandwidth for peak concurrency
The advertised codec bit rate is not the complete network requirement because IP, UDP, RTP, Ethernet, and security overhead add data. Capacity should be calculated per call in both directions, multiplied by peak concurrent calls, and increased with safety headroom. Stable latency, jitter, and packet loss matter more than a high speed-test result alone.