Asynchronous Transfer Mode (ATM) is a protocol that is different from VoIP. This article will explain the difference, and why VoIP is better.

“Asynchronous” means that something is not synchronized; in telecommunications, data transmission can be (but in this case is not) synchronized to a clock signal. In this way, ATM is not circuit-switched, but it is designed to create a virtual circuit over a packet-switched network. ATM connections are made through fiber or twisted pair cable, usually a phone or ISDN cable for the end-user.

Asynchronous Transfer Mode uses time-division multiplexing, a method of encoding data into packets. The packets, also called cells, are all the same size, which is 48 bytes of data and a five byte header, making it 53 bytes total. Based on the timing, the data is reassembled at the termination point. Specialized hardware called “ATM switches” establish a point-to-point communication, and then these packets of 53 bytes each are then sent in order, and sent together. They can be sent at a constant or variable bitrate, but the size of the bits must be the same.

ATM was designed to maintain Quality of Service (QoS), meaning that the data would be transmitted at a constant speed, reducing jitter. ATM traffic can transmit voice, video, and data signals, much like VoIP. The ATM switch is part of the PSTN and a large organization can have its own private ATM network. With the ATM switch, the ISP could send a large amount of data at once (over 600 Mbps), while ensuring that no single transmission hogs up all the bandwidth, ensuring that everyone gets good service. The underlying assumption, as the ATM protocol was being developed in the early 1990s, was that bandwidth was a limited resource.

Compared to VoIP, the biggest drawbacks of ATM are that ATM packets are all the same size, and must be sent along the same path. VoIP packets are a minimum of 218 bytes each, but can be compressed to as low as 2 bytes each. VoIP codecs allow for variations in size that Asynchronous Transfer Mode does not. VoIP packets contain header information about the destination and the order in which they are to re-encoded, so the packets can travel through the best available path at the best speed. In practice, this leads to lower (cheaper) bandwidth requirements. VoIP does not require the end user to have any specialized equipment. VoIP traffic goes through the same cables to the same internet that you are already using.

Telecommunications experts once predicted that Asynchronous Transfer Mode would be the future of the internet, but it is ultimately turning into a product designed for the 20th century. Voice over Internet Protocol has inherent load-balancing properties that make ATM unnecessary, and VoIP providers user a combination of hardware and customer service to ensure proper Quality of Service. Limited bandwidth is becoming a thing of the past, as ISPs are constantly innovating new ways of delivering screamingly fast bandwidth. That bandwidth is being filled by 21st century innovations like telepresence and HD audio.


Additional Reading

VoIP Jitter & Latency: Causes and How to Troubleshoot
VoIP vs Landline In Depth Comparison