Speech transmission in a packet network

Multiplex communications – Pathfinding or routing – Switching a message which includes an address header

Reexamination Certificate

Rate now

  [ 0.00 ] – not rated yet Voters 0   Comments 0

Details

C370S395100

Reexamination Certificate

active

06738374

ABSTRACT:

This application also claims the national phase of international application PCT/FI97/00194 filed Mar. 27, 1997 which designated the U.S.
FIELD OF THE INVENTION
The invention relates to speech transmission in a packet network and especially to transmission between a transcoder and a base station of a digital mobile communication network.
The invention will be explained in connection with speech processing and speech frames but the same technique can be applied to transmission of a music and video signal. It is common to these signals that signal samples have to be conducted isochronously to a decoder, that is, essentially at intervals equal to the intervals at which the samples are formed in the encoder.
BACKGROUND ART
In a digital telephone system a speech signal is encoded in some manner before it is channel coded and sent to the radio path. For example, in the case of the GSM system, digitalized speech is processed frame by frame at intervals of about 20 ms by using different methods so that it results in a parameter group representing speech for each frame. This information, that is, the parameter group is channel coded and sent to the transmission path. The used speech coding algorithms are RPE-LTP (Regular Pulse Excitation LPC with Long Term Prediction) and various code excited algorithms CELP (Code Excited Linear Prediction) of which VSELP (Vector-Sum Excited Linear Prediction) should be mentioned.
In addition to actual coding, the following functions are also built in for speech processing: a) on the transmitter side Voice Activity Detection VAD with which the transmitter can be instructed to be switched on only when there is speech to be sent (Discontinuous Transmission, DTX), b) on the transmitter side the evaluation of background noise and the generation of respective noise parameters and on the reception side the generation of comfort noise in a decoder from the parameters, and c) acoustic echo suppression. Noise during a break makes the connection sound more pleasant than absolute silence.
In a known GSM mobile telephone system the input of a speech encoder is either a PCM signal of 13 bits from the network or an A/D converted PCM of 13 bits from the audio part of the mobile station. The speech frame obtained from the output of the encoder is 20 ms in duration and comprises 260 audio bits which are formed by encoding 160 PCM-encoded speech samples. Voice Activity Detection (VAD) defines from the parameters in the speech frame whether or not the frame contains speech. If speech is detected, the frames transmitted to the radio path as so-called traffic frames are speech frames. After a speech burst, and at specified intervals also during speech pauses indicated by the VAD, the traffic frames are SID frames (Silence Descriptor) containing noise parameters, in which case the receiver is able to generate from these parameters noise similar to the original noise also during pauses.
A traffic frame thus contains a speech block of 260 bits representing 20 ms of encoded speech/data or noise. Furthermore, the frame has 56 bits available for frame synchronization, speech and data indication, timing and other information, the total length of the traffic frame being 316 bits. Uplink and downlink traffic frames differ slightly from one another in these 56 bits.
Referring to
FIG. 1
, which shows a simplified view of the present GSM network from the point of view of transmission. Network Subsystem comprises a mobile service switching centre, the mobile communication network being connected via the system interface of the mobile services switching centre to other networks, such as Public Switched Telephone Network PSTN. Via A interface the network subsystem is connected to the base station subsystem BSS comprising base station controllers BSC and base stations BTS connected thereto. The interface between the base station controller and the base stations connected thereto is an Abis interface. The base stations are in radio communication with mobile stations via the radio interface. Traffic frame forming unit TRAU explained above is in the figure placed in association with the base station but it may also be situated in association with the mobile services switching centre.
The mobile services switching centre MSC is shown in a simplified way in FIG.
2
. Control of the base station system BSS is one function of the mobile services switching centre in addition to a call control. The function of the switching matrix is to select, switch and separate speech/data and signalling paths passing through it in a desired way. The switching matrix switches in this way its part of the connection between a mobile subscriber and a subscriber of another network or of the connection between two mobile subscribers. The function of the Network Interworking Functions IWF
1
is to adapt the GSM network into other networks. The PCM trunk line is connected to a PBX system by a terminal circuit trunk interface
3
so that the physical interface of layer
1
between the exchange and the base station controller BSC is a line of 2 Mbit/s, that is, 32 time slots of 64 kbit/s (=2048 kbit/s). The signalling terminal
4
carries out signalling according recommendation CCITT No:
7
.
The functions of the base station controller BSC indicated with reference
14
in
FIG. 1
include selection of a channel between it and the mobile station, link control and channel release. It carries out mapping from the radio channel to the channel of the PCM time slot of the interconnecting line between the base station and the base station controller. The base station controller shown in a simplified way in
FIG. 3
comprises terminal circuits, trunk interfaces
31
and
32
by means of which the base station controller is connected on the one hand to the mobile services switching centre over the A interface and on the other hand to the base stations over the Abis interface. Transcoder and Rate Adaptation Unit TRAU is an element of the base station system BSS and it may be situated in association with the base station controller BSC as shown in
FIG. 1
, or also in association with the mobile services switching centre, for example. The transcoders convert speech from one digital format to another, for example, they convert the 64 kbit/s A-law PCM from the exchange over the A interface into encoded speech of 13 kbits to be sent to the base station line and vice versa. Rate adaptation for data is carried out between the rate 64 kbits and the rates 3.6, 6 or 12 kbit/s.
The base station controller BSC configures, allocates and supervises the circuits of 64 kbit/s in the direction of the base station. It also controls the switching circuits of the base station by means of the PCM signalling link and allows the circuits of 64 kbit/s to be used efficiently, that is, a switch at the base station, which the base station controller controls, switches transmitter/receivers to PCM links. This switch hence operates as a drop/insert multiplexer, i.e. as an add/drop multiplexer which drops a PCM time slot for the transmitter of the data or inserts a reception time slot to a PCM time slot of the data or links the PCM time slots forwards to other base stations. The base station controller thus sets up and releases connections to the mobile station. The connections from the base stations to the PCM line or lines over the A interface and the procedure in the opposite way are multiplexed in a switching matrix
33
.
The physical interface of layer
1
between the base station BTS and the base station controller BSC is a line of 2 Mbit/s, that is, 32 time slots of 64 kbit/s (=2048 kbit/s). The base station is totally controlled by the base station controller BSC and it mainly contains transmitter/receivers TRX which implement the radio interface towards the mobile station. Four full rate traffic channels via the radio interface can be multiplexed into one PCM channel of 64 kbit/s between the base station controller and the base station, in which case the rate of the speech/data channel is in this interval 16 kbit/s. In that case, on

LandOfFree

Say what you really think

Search LandOfFree.com for the USA inventors and patents. Rate them and share your experience with other people.

Rating

Speech transmission in a packet network does not yet have a rating. At this time, there are no reviews or comments for this patent.

If you have personal experience with Speech transmission in a packet network, we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Speech transmission in a packet network will most certainly appreciate the feedback.

Rate now

     

Profile ID: LFUS-PAI-O-3208178

  Search
All data on this website is collected from public sources. Our data reflects the most accurate information available at the time of publication.