How VoIP Works - III

 

VoIP: Codecs

A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of VoIP.
Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:

• 64,000 times per second
• 32,000 times per second
• 8,000 times per second

A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.
Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states "if no one is talking, don't send any data." The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It's Annex B in the CS-ACELP algorithm that's responsible for that aspect of the VoIP call.
The codec works with the algorithm to convert and sort everything out, but it's not any good without knowing where to send the data. In VoIP, that task is handled by soft switches.
E.164 is the name given to the standard for the North American Numbering Plan (NANP). This is the numbering system that phone networks use to know where to route a call based on the dialed numbers. A phone number is like an address:

(313) 555-1212 313 = State 555 = City 1212 = Street address

The switches use "313" to route the phone call to the area code's region. The "555" prefix sends the call to a central office, and the network routes the call using the last four digits, which are associated with a specific location. Based on that system, no matter where you're in the world, the number combination "(313) 555" always puts you in the same central office, which has a switch that knows which phone is associated with "1212."
The challenge with VoIP is that IP-based networks don't read phone numbers based on NANP. They look for IP addresses, which look like this:

192.158.10.7

IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway or a telephone. However, IP addresses are not always static. They're assigned by a DHCP server on the network and change with each new connection. VoIP's challenge is translating NANP phone numbers to IP addresses and then finding out the current IP address of the requested number. This mapping process is handled by a central call processor running a soft switch.

The central call processor is hardware that runs a specialized database/mapping program called a soft switch. Think of the user and the phone or computer as one package -- man and machine. That package is called the endpoint. The soft switch connects endpoints.
Soft switches know:

• Where the network's endpoint is
  
• What phone number is associated with that endpoint
• The endpoint's current IP address

VoIP: Soft Switches and Protocols

 The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.

Protocols
As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls. How do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is protocols.
There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols, H.323 incorporates many individual protocols that have been developed for specific applications.

H.323 Protocol Suite
Video	Audio	Data	Transport
H.261 H.263	G.711 G.722 G.723.1 G.728 G.729	T.122 T.124 T.125 T.126 T.127	H.225 H.235 H.245 H.450.1 H.450.2 H.450.3 RTP X.224.0

As you can see, H.323 is a large collection of protocols and specifications. That's what allows it to be used for so many applications. The problem with H.323 is that it's not specifically tailored to VoIP.
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waiting. You can learn more about the architecture of these protocols at Protocols.com: Voice Over IP.

One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.

VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it's clear that developers will keep refining this technology until it eventually replaces the current phone system.

On the next article, we'll talk about VoIP call monitoring.