CompTIA Exam Prep - ITF+, A+, Network+, Security+, CySA+

SIP sets up, changes and tears down media sessions. It does not carry the audio or video itself — that is RTP's job. Keeping those two separate is the single most useful thing to understand about voice over IP, because almost every troubleshooting question and every security question turns on it. Signalling and Media Are Different Streams SIP negotiates. RTP delivers. They use different ports, frequently take different paths through the network, and fail independently. That independence produces the classic symptom: the phone rings, someone answers, and neither side hears anything. Signalling worked, media did not. The fault is almost always a firewall or NAT device permitting SIP on port 5060 while dropping the dynamically negotiated RTP ports. The Call Flow A basic call is worth memorizing as a sequence: INVITE — the caller requests a session, carrying an SDP body describing the codecs and the address it wants media sent to. 100 Trying — the next hop ...

Quality of Service does not create bandwidth. It decides who suffers when there is not enough. Every QoS design is a statement about which traffic the business is willing to drop or delay, and the technical machinery exists only to carry out that decision reliably. What Traffic Actually Needs Three measurements, and different applications care about different ones: Bandwidth — how much capacity. A file transfer wants a lot and does not care when it gets it. Latency — one-way delay. Voice needs under about 150 milliseconds before conversation becomes awkward. Jitter — variation in latency. Voice and video tolerate a consistent delay far better than an inconsistent one, because a jitter buffer can absorb the former. Loss matters too, but differently: a retransmitted file is fine, a retransmitted voice packet is useless because it arrives after its moment. The Four Steps Classification identifies traffic — by port, protocol, source, or deep inspe...