VoIP
Technology
The VoIP technology requires that the voice should be compressed, by means of software algorithms or DSP (Digital Signal Processor), fragmented and inserted in IP packets.
Voice encoding
There are several ways of encoding the voice (or "codecs"), that are different for the ratio of compression of the voice data. Of course the higher the compression the lower the bandwidth used, but also the lower the quality of the voice. The most used codecs are G.711 at 64 kb/s, G.729 at 8 kb/s (probably the best bandwidth/quality trade-off) and G.723.1 at 6.3 or 5.3 kb/s.
Protocols
The are also several ways of communications inside the VoIP technology. The main protocols that carry the voice packets are H.323, SIP (Session Initiation Protocol) and MGCP (Media Gateway Control Protocol). Usually these standards require different channels for the voice transmission and for the signalling of the communication. For voice transmission the standard protocol is RTP (Real-time Transport Protocol) based on UDP. For the signalling channel the standard protocol is TCP.
The following table shows the relation between the OSI model (Open System Interconnection) and the protocols used for VoIP.
| OSI Level | VoIP protocols | |
|---|---|---|
| 7 | Application | NetMeeting/GnomeMeeting/Applications |
| 6 | Presentation | Codecs |
| 5 | Session | H.323/MGCP/SIP |
| 4 | Transport | RTP/TCP/UDP |
| 3 | Network | IP |
| 2 | Data Link | Frame Relay, ATM, Ethernet, PPP, MLP, and others |
