The Transmission Performance of Bell System Toll Connecting Trunks

To characterize the transmission performance of the Bell System switched telecommunications network, Bell Telephone Laboratories conducted a survey of toll connections during 1969 and 1970. Connections were established between Bell System end offices chosen by statistical sampling techniques. Both analog and data transmission tests were performed. A summary of analog transmission performance is presented in this paper. It contains estimates for noise, loss, attenuation distortion, envelope delay distortion, peak‐to‐average ratio, frequency offset, level tracking, nonlinear distortion, and phase jitter for toll calls within the Bell System. Accompanying papers discuss data transmission error performance at various speeds between 150 and 4800 bits per second.

1969-70 Connection Survey: Analog Transmission Performance on the Switched Telecommunications Network

Duffy¹,

Thatcher²

1971

Loss-Noise-Echo Study of the Direct Distance Dialing Network

Spang¹

1976

A review of the current VIA Net Loss plan was made using a simu lation model of the transmission characteristics of telephone connections. This model allowed the determination of the optimal loss which balanced the degradation due to talker echo and the degradation caused by its control on primary speech. The review reaffirmed current methods with some changes to allow for the emerging digital network.

A Subjective Comparison of Selected Digital Codecs for Speech

Daumer¹,

Cavanaugh²

1978

Technological advances are continually increasing the economic viability of efficient codecs in telephone networks. A subjective evaluation is described here of the μ255 Pulse Code Modulation (PCM) algorithm and three more efficient techniques, Nearly Instantaneous Companding pcm (nic pcm), Cummiskey‐Jayant‐Flanagan Adaptive Differential pcm (adpcm), and Subscriber Loop Carrier Adaptive Delta Modulation (SLC* adm). These codecs are compared under the conditions of: (i) single encodings as a function of line bit rate, input level, received volume, and error rate, (ii) tandem encodings with intermediate baseband conversion, and (iii) local, exchange, and toll network reference connections where mixed tandem encodings might be found along with typical analog impairments such as loss and random noise. The simulation of the codec algorithms on a minicomputer facility enabled the production of subjective test tapes containing speech processed under these conditions. These tapes were then evaluated in listening‐type subjective tests. It is shown that (i) nic pcm, adpcm, and SLC adm have approximately a 12‐ to 16‐kb/s advantage over μ255 pcm for equivalent subjective ratings, (ii) nic pcm, adpcm, and SLC adm perform comparably over the range of conditions tested; and (iii) 64‐kb/s μ255 pcm can be deployed in a multiple encoding environment with very few restrictions, whereas the use of lower bit rate nic pcm, adpcm, and SLC adm codecs would necessitate more stringent application rules to avoid excessive degradation in tandem encoding situations.