VoIP gateway(1)

VoIP gateway is a piece of hardware with the standard purpose of converting TDM telephony traffic from the PSTN into digital packets IP packets for transport over an IP network (such as your LAN). A VoIP gateway can also convert digital IP packets into TDM telephony traffic for transport across the PSTN (Publicly Switched Telephone Network).

How does a VoIP gateway work?
VoIP gateways are rather simple. A VoIP gateway works as a bridge between an IP network and the PSTN. Depending on where the voice traffic originates from a VoIP gateway will convert the voice traffic into the proper form for receipt by the destination network (IP or PSTN).

If the voice traffic is originating from the PSTN the VoIP gateway will convert the analog voice signal into a digital signal. This digital signal is then compressed using a codec and broken into a series of packets that are transferred across the IP network using a signaling protocol.

If the voice traffic is originating from an IP network the VoIP gateway will decompress the digital packets into a digital signal that is then converted into an analog signal to be sent across the PSTN.

VoIP gateway protocols and codecs
Hopefully your head’s not spinning from the explanation of how a VoIP gateway works. Because it’s now time to talk about VoIP protocols and voice codecs.

Even if you’re not technical you still need a basic understanding of VoIP protocols and codecs. In order for your gateway to work properly it must use a protocol and codec that is compatible with your VoIP phone system and or VoIP service. The protocol and codec you use can also drastically increase or decrease the quality of your calls.

VoIP protocols
A VoIP protocol determines how your voice packet is transported across a network. A VoIP gateway will typically support a single protocol.

The most common VoIP protocols are:

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.
SCCP (Cisco Skinny Client Control Protocol) – SCCP is a proprietary protocol used by Cisco’s Call Manager and IP phones.
MGCP – MGCP is an older VoIP protocol you might come across. It is no longer widely used and or supported.
H.323 – Similar to MGCP, H.323 is an older VoIP protocol that you might come across, but is no longer widely used and or supported.
Voice codecs
A voice codec is responsible for the compression of your voice stream within a digital packet. It also determines sound quality and bandwidth required to send the packet. A VoIP gateway 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
If you’re the person responsible for the set-up, installation and maintenance of a VoIP gateway you will want to further your knowledge in the area of protocols and codecs. If not, simply make sure your VoIP gateway supports the same protocols and codecs that your VoIP service and or VoIP phone system support.