VoIP protocols and codecs
A VoIP phone system connects to your Local Area Network (LAN) and uses it as the backbone of your system. When you connect your VoIP phones and your VoIP service provider to the VoIP PBX, you’ll probably use HD IP phones to communicate. A VoIP phone system uses IP technology to handle your call control and manage your connections to the Wide Area Network. Even though a VoIP phone system uses VoIP and is connected to your LAN, most systems can connect directly to the Publicly Switched Telephone Network (PSTN). This gives you the ability to use both VoIP and the PSTN for your calling.
A VoIP protocol determines how your voice packet is transported across a network. A VoIP phone will typically support one protocol.
One of the most popular VoIP protocols is:
SIP (Session Initiation Protocol) – SIP is a standards-based protocol that is used and supported by the vast majority of VoIP phone systems and services.
A voice codec is responsible for the conversion of your analog voice stream into a digital packet. Voice codecs also determine sound quality and bandwidth required to send the packet. A VoIP phone typically supports multiple voice CODECs.
The most common voice codecs are:
- GSM – 13 Kbps
- iLBC – 15 Kbps
- G.711 – 64 Kbps
- G.722 – 48/56/64 Kbps
- G.726 – 16/24/32/40 Kbps
- G.728 – 16 Kbps
- G.729 – 8 Kbps