Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching
Reexamination Certificate
1997-09-17
2001-01-09
Kizou, Hassan (Department: 2738)
Multiplex communications
Pathfinding or routing
Combined circuit switching and packet switching
C370S356000, C370S392000, C370S397000
Reexamination Certificate
active
06172973
ABSTRACT:
BACKGROUND OF THE INVENTION
Field of the Invention
The present invention relates to telecommunications in general and in particular to sending voice over high speed digital communication circuits.
DESCRIPTION OF THE RELATED ART
FIG. 1
illustrates a typical network with voice over ATM capability. Since voice switching is done in a time division multiplexing (TDM) domain, the ATM background is used only as a transport network. At least two TDM voice switches will typically be connected to the ATM backbone network at each originating and terminating end. TDM voice switches use 64 KB per second samples of digitized voice for it's switching fabric. Once the switching is done these voice samples are packetized into ATM cells which are 53 bytes long before sending them through the ATM network. On the terminating side, the ATM cells are again depacketized and converted into voice samples before entering the TDM voice switch.
During each portion of the transmission, the digitized TDM voice signal is packetized and de-packetized. Each portion of the network traveled introduces its own delay. These delays, while insignificant in communicating standard data take on greater significance when voice comes into play. Delays in voice conversations soon become unacceptable and are unpleasant to the ear.
There are several delays to be considered. The cell fill delay is the delay required to fill a 53 byte ATM cell with 64 KBPS TDM voice samples. It requires about 6 ms (376 bits divided by 64,000 bps) to fill an ATM cell with TDM voice samples, assuming one cell is used per voice call. This delay increases as voice compression is used in the network. In fact, the use of compression increases the amount of delay by a factor of the compression ratio. For example, if the voice is compressed at a 4 to 1 ratio, which is typically used, then it would increase the delay by a factor of four which is 4 by 6 ms equals 24 ms at the originating end.
On the terminating side, another delay is introduced to allow all the ATM cells to arrive into a smoothing out buffer before playing the voice. The purpose of the smoothing out is to eliminate the probability of a cell not being present at the time of playing of the voice. This delay is sometimes called cell playout delay and is calculated to be less than 7 ms for both 4 to 1 compression networks and for 8 to 1 compression networks.
The use of voice compression in the network adds to yet another end to end delay. The most commonly used 4 to 1 compression by adaptive differential pulse code modulation (ADPCM ) technology typically involves a delay of 25 ms. A new technology called low delay-code excited linear prediction (LD-CELP), incurs the delay of 2 ms for 8 to 1 compression. All things considered, a minimum delay of 56 ms for 4 to 1 compression to 57 ms for 8 to 1 compression is added for a simple network with 2 voice switches as shown in FIG.
1
. Reality is however, that most voice calls will go through at least 4 or more voice switches at multiple entry and exit points to and from the ATM backbone network. These entries and exit will introduce significantly more delay to the process.
FIG. 2
illustrates an example of a network with 4 TDM voice switches and the respective interfaces to ATM networks. In this scenario voice packets interact with 3 separate ATM networks. As mentioned above each time 2 voice switches interact with an ATM network, a delay of 56 to 57 ms is added to the voice transmission. The network illustrated in
FIG. 2
will incur a delay of 168 to 171 ms depending on the voice compression factor (4 to 1 or 8 to 1). The acceptable delay for the real-time voice transmission is required to be less than 130 to 200 ms. Any greater delay results in very unpleasant conversation between the end users. The increased frequency of compression and decompression makes matters worse by introducing errors in the voice signal. It is therefore desirable to eliminate and/or reduce the number of exit and entry points between the voice network and the ATM network.
There is accordingly a need for a new method and apparatus for voice communication over data networks in order to solve or ameliorate one or more of the above-described problems.
SUMMARY OF THE INVENTION
The instant invention allows control of voice transmission from end user to end user by common channel signaling without the repeated conversion delays encountered by entering and exiting the standard voice network and an ATM network. The control is accomplished in parallel over the standard public switch telephone network while the voice “data” is transported over an ATM “data” network. Each ATM node has with it a co-located PSTN voice switch. A pair of control links are used to communicate between the two co-located ATM and TDM switches.
Further features of the above-described invention will become apparent from the detailed description hereinafter.
The foregoing features together with certain other features described hereinafter enable the overall system to have properties differing not just by a matter of degree from any related art, but offering an order of magnitude more efficient use of processing time and resources.
Additional features and advantages of the invention will be set forth in part in the description which follows, and in part will be apparent from the description, or may be learned by practice of the invention. The advantages of the invention will be realized and attained by means of the elements and combinations particularly pointed out in the appended claims.
It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory only and are not restrictive of the invention, as claimed.
The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate preferred embodiments of the apparatus and method according to the invention and, together with the description, serve to explain the principles of the invention.
REFERENCES:
patent: 5204857 (1993-04-01), Obara
patent: 5483527 (1996-01-01), Doshi et al.
patent: 5526353 (1996-06-01), Henley et al.
patent: 5600641 (1997-02-01), Duault et al.
patent: 5734653 (1998-03-01), Hiraiwa et al.
patent: 5764644 (1998-06-01), Miska et al.
patent: 5862136 (1999-01-01), Irwin
patent: 5889765 (1999-03-01), Gibbs
patent: 5946323 (1999-08-01), Eakins et al.
patent: 5956334 (1999-09-01), Chu et al.
patent: 6014378 (2000-01-01), Christrie et al.
patent: 6067299 (2000-05-01), DuRee
patent: 0 711 052 A1 (1996-05-01), None
patent: 2 282 027 (1994-08-01), None
patent: WO 94/11975 (1994-05-01), None
patent: WO 97/03526 (1997-01-01), None
patent: WO 97/16005 (1997-05-01), None
P Heywood; Data Communication, vol. 25, No. 16, Nov. 21, 1996, pp. 33-34, XP002087865 us see p. 34, line 7-line 11; figure 1.
Akhtar Haseeb
Akhtar Shahid
Haynes & Boone LLP
Kizou Hassan
Nortel Networks Limited
Pezzlo John
LandOfFree
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