Jade Clayton's Pages at privateline.com (1) (2)

The Third Edition definition of TDMA on page 648 was messed up, no doubt the result of deadline pressure. Jade and I are going to come up with a proper definition soon, for now I have prepared a cartoon. See if this makes sense:

Each face represents a conversation or a part of a conversation. With FDMA we put different calls on different frequencies, like broadcast stations are separated or divided by frequency. You know, A.M. station 560, 570, 580, 590, 600, 610 and so on. With time division multiple access we divide each call on a single frequency by time, like talking in turn. With CDMA we assign an identifying code to each call and put bits and pieces of different calls on different frequencies as the conversation continues. AT&T's national wireless network, as well as GSM, use TDMA. Sprint's PCS network uses CDMA. I have more on multiplexing and the like throughout this site.

The Skinny on IP Telephony by Jade Clayton

There are some basic concepts that traditional minded telecommunications enthusiasts need to grasp before embarking on the transition to packet based telephone networks (link to Tom Farley's writing). The most important is that the telephone network is not just a telephone network anymore. Beyond that, understanding the different challenges that telecommunications has always faced, and the best way to overcome them. The biggest problem with traditional TDM (Time Division Multiplexed) networks, or circuit switched networks (link to Tom Farley's writing) is that they are difficult to scale, and difficult to make fault tolerant in comparison to packet switched networks. This especially includes ATM (Asynchronous Transfer Mode), which has been called a "cell" switching technology. In reality, the kernel of ATM technology is circuit based. It has been disguised with terminology as a "fixed length packet" or cell. In the near future ATM will only have new deployments in large telecommunications carrier networks where bandwidth is sold in tangible chunks. In the medium term, ATM will be completely phased out by less expensive, more nimble, and much faster packet based networks.

What about telephone switching? When one thinks of the traditional telephone call, the DS0 comes to mind. The DS0 is a TDM channel that can fit into a 24 Channel T1, which can fit into a 28 channel T3, then be transported over an STS-1(definition) or an STM-1 depending on the country in which you live. [See the table below] Finally at the top of the hierarchy, we have the SONET transport method, which stacks these large packages on top of each other to obtain more and more individualized bandwidth. This hierarchy will be a memory in a few years to come.

So what is to become of the 64 Kbps DS0? The DS0 will become a G.711 packet stream (definition) within a packet network, specifically 802.3 flavored Ethernet, which can now run at speeds of 10Gbps. When the G.711 packet stream enters an IP network, it is managed by QoS [Quality of Service] enhanced routers (or Layer 3 Switches) which are able to identify and prioritize the IP packets according to Voice, Video, or Data. Many systems are capable of compressing a G.711 packet stream into a G.729 packet stream (definition), which only consumes 8 Kbps of bandwidth without overhead. This easily enables over one hundred telephone calls over a traditional T1 (1.544 Mbps) bandwidth. With traditional telephone methods, only twenty four calls could traverse a T1.

So how does dial tone, and other signaling take place? There are several protocols that can perform this operation over a packet network. MGCP (Media Gateway Control Protocol) is the most adopted standard at this time. SSP (Cisco's Skinny Station Protocol) is also used, and most likely to be the most widely adopted. MGCP is used where IP Telephony integrates with traditional telephone methods, and SSP is utilized in end-to-end IP Telephony connections. There are other protocols used to accomplish real-time voice connectivity.

One of the great things about IP Telephony is that Voice is now an application that runs over a data network. The voice trunks and connections are managed by a central server. Servers can be implemented redundantly in clusters, and be equipped with internal hard drive redundancy. This exceeds the fault tolerance of traditional telephone circuit switches by several orders of magnitude, and costs about one-fourth to implement.

IP Telephony is the stepping stone to voice/internet integration, and the prelude to the wide deployment and use of Video-Phone.

STS-1: The basic bandwidth building block SDH or Synchronous Digital Hierarchy. The STM-1 signal has a bandwidth of 155.52 Mbps. . . Identical to the electrical STS-1 amd the optical OC-1 formats.

G.711. A codec standard. The G.711 ITU-T standard for voice compression describes the 64Kbps PCM voice coding technique. In G.711, encoded voice is already in the correct format for digital voice delivery in the Public Switched Telephone Network or through PBXs. Using this type of compression an a VoIP network reduces delay and improves voice quality. It also reduces the loads placed on routers performing the digital signal processing. The disadvantage is that is it is a "bandwidth guzzler" in terms of connections over a wide area network link. The bandwidth per conversation required, including Ethernet and other framing overhead, can approach 128 Kbps. The most important aspect of G.711 and other Codec methods is that they provide a true convergence of traffic for VoIP networks, which makes for better end-user management of bandwidth.

G.729: Another codec or compression standard. The G.729 ITU-T standard describes CELP (Code Excited Linear Prediction) where voice channels are coded into 8Kbps streams [Something like the bandwidth in a Real Audio stream, ed.] . . . The G.729 standard is preferred in a Voice over Internet Protocol (VoIP) network because it conserves bandwidth on an 8 to 1 ratio. . ."

SONET: Broadband transport method designed exclusively for fiber optic cable. . . Based on a hierarchy of STS or Synchronous Transport Signals [which have] a transport speed of 51.84 Mbs. .. The important thing to know about a SONET network is that it simply replaces the older telecommunications technology copper twisted pair outside plant with fiber optic and electronics.

Jade Clayton writes about what is happening now with voice over the internet. For background and a more lengthy explanation, click here for a selection from the McGraw Hill Broadband Telecommunications Handbook by Regis J. Bates (14 pages, 315K in .pdf)

For yet another explanation on Voice over IP, click here for a selection from Carrier Grade Voice Over IP by Daniel Collins (20 pages, 860K in .pdf)