VoIP - Anatomy of a voice communication
A typical voice communication using Voice over IP (VoIP) technology has two components: the signaling component, and the actual voice component.
The voice component carries voice packets from the source device to the destination. These packets actually contain the digitized and packetized voice of the people talking on the phone.
The signaling component deals with setting up the call, maintaining the voice session and tearing it down once the call is complete. Signaling is also involved with transmitting things like dual tone multi frequency (DTMF) dialing, known as "touch tone dialing" and also controls the various telephony features and tones, such as making a phone ring, playing back busy tone, and implementing typical telephony features call waiting, call transfer, call hold, and conferencing, to name a few.
The signaling session is separate and distinct from the voice packet session, although these two sessions are related, and operate together to successfully complete a telephone call.
The most common protocol used for signaling in VoIP today is Session Initiation Protocol (SIP). However, others such as H.323 are also widespread.
Links to this page:
- Carrier-Grade NAT
- Hardware - Application Specific Integrated Circuit (ASIC)
- Interface - show interfaces counters explained
- QoS - Jitter
- QoS - classification
- Real-time Transport Protocol
- Routing - Cisco Performance Routing (PfR)
- Security - spoofing
- Session Border Controller
- Session Initiation Protocol
- Wireless Fast BSS Transition
- Wireless roaming defined