Multiplex communications – Communication techniques for information carried in plural... – Assembly or disassembly of messages having address headers
Reexamination Certificate
1999-05-07
2004-04-06
Chin, Wellington (Department: 2664)
Multiplex communications
Communication techniques for information carried in plural...
Assembly or disassembly of messages having address headers
C370S476000
Reexamination Certificate
active
06717955
ABSTRACT:
BACKGROUND
This invention relates generally to the packetization of data streams. More particularly, this invention relates to packetizing voice signals and transporting the packetized voice signals using Asynchronous Transfer Mode (ATM) after having multiplexed the packetized voice signals from several sources into the same ATM cell.
Voice or speech information in existing telecommunication networks, as e.g. Public Switched Telephone Network (PSTN) and Integrated Services Digital Network (ISDN), is normally transported as PCM coded samples. According to ITU-T G.711, PCM stands for Pulse Code Modulation of voice frequencies and means that the basic time slot is 8 bits every 125 &mgr;s, giving rise to channels of 64 kbit's. Consequently, PSTN and ISDN are 64 kbit/s circuit-switched networks. Circuit stands for a set of physical transmission resources, e.g. lines and exchanges, that provide for a two-way transfer of message signals from source to destination in a telecommunication system. Several circuits are time multiplexed over one link by joining together several time slots in a frame which is again repeated with a certain frequency. A circuit will always use the same time slot in the frame during the complete duration of the session. When transporting the PCM coded samples over an ATM network, i.e. a packet switching network instead of a circuit-switched data network, one way for the ATM network to achieve this is to emulate a virtual circuit. The emulation is carried out by a Circuit Emulation Service (CES) as specified by the ATM Forum Circuit Emulation Service Interoperability Specification Version 2.0, which comprises an ATM Adaptation Layer 1 (AAL1) adapting circuits of constant bit rate to an ATM cell transport.
Transporting a 64 kbit/s circuit using AAL1 can be done in basically two ways. One is to entirely fill the ATM cell payload, i.e. the information field, generally of 47 octets (48 octets for the information field-1 octet for the AAL1 header) with PCM samples, resulting in an ATM packetization delay of 47*125 &mgr;s, i.e. 5875 ms. The extra overhead added by the 5 octets ATM cell header and 1 octet AAL1 packet header amounts to roughly 13 percent of the nominal bit rate of 64 kbit/s. The other way is to only partially fill the ATM cell payload with PCM samples before sending it, and thus reduce the ATM cell packetization delay, but paying penalty by increased overhead. For instance, by only filling the cell by a sixth, i.e. by around 8 PCM samples, the delay can be reduced to around 1 ms. It is important to keep delay at a low level for all parts of the network, since delay in combination with echo may severely degrade the perceived voice quality.
Voice signals may also be carried in other encoding formats than the above mentioned PCM format. These encoding formats being a part of a voice codec may be according to e.g. the Global System for Mobile Telecommunication (GSM) standards, Adaptive Differential PCM (ADPCM) standards, Low Delay Code Excited Linear Prediction (LD-CELP) standards or Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) standards. A common property for all these voice codes is that they compress the voice signals according to a compression algorithm such that they produce coded voice information, i.e. packets or frames, with a resulting bit rate which is lower than the common 64 kbit/s PCM based coding scheme. The result is therefore, that when using AAL1, the ATM packetization delay and/or overhead (if partially filled cells are used) increases roughly by a factor equal to the degree of compression. For instance, an 8 kbit/s CS-ACELP coded voice signal would result in an ATM packetization delay as long as 47 ms. This could as before be traded by partially filling the cell, bit with the same scaling factor of 8 for the resulting bandwidth overhead. In addition, some voice codes based on GSM and CS-ACELP already have an inherent algorithm delay which is fairly large which also has to be added in a total network perspective.
The different voice codes produce packets at varying rates and sizes which are not commensurable with the ATM payload length, resulting in padding or segmentation problems. With these voice encoders it is also possible to filter out silent periods of speech and to reduce or stop the emission of voice data during those silent periods (hereinafter referred to as silence removal). Silence removal results in a variable bit rate, which is not suitable for a constant bit rate transport service such as the CES based on AAL1.
A new ATM Adaptation Layer (AAL) for the purpose of transporting packetized low bit rate data, such as packetized voice, and which data is real-time critical and of variable rate and length, is in the process of being standardized. This new AAL is denoted AAL type 2 (AAL2), which is specified in the ITU-T Draft 1.363.2, B-ISDN ATM Adaptation Layer Type 2 Specification. Seoul, February 1997. AAL2 is designed to provide support for applications requiring low delay with reasonable bandwidth efficiency and variable packet sizes. AAL2 is also asynchronous.
In AAL2, voice signals from several sources are multiplexed into the same ATM connection. This is done by encapsulating user data into AAL2 packets, which are multiplexed using the AAL2 multiplexer function (AAL2 MUX) inherent in the AAL2 Common Part Sublayer, into an ATM connection. Reduced delay is provided through inclusion of a Combined Use-timer (Timer_CU) in the AAL2 which guarantees a maximum holding time of user data, or AAL2 packets, before transmission of the carrying ATM cell. The maximum packetization delay introduced by the AAL2 ATM is then basically equal to the value of the Timer_CU. The penalty to be paid is overhead in bandwidth if an ATM cell is not completely filled upon transmission.
In a typical scenario using AAL2, there are n voice sources which ire multiplexed by an AAL2 MUX into the same ATM connection after having been processed by an encoder with a suitable voice coding algorithm. Each voice encoder may be combined with a packetizer such that the AAL2 MUX is presented with packets of a suitable periodicity. The voice encoders may also include means for silence removal. Hence, after processing, the AAL2 MUX is presented with packets having a variable length p and/or packet rate f. A typical appropriate value for T
Timer
—CU
is 1 ms, which is based on calculations and simulations. Generally, T
Timer
—CU
is much less than the packet periodicity t (=1/f) for most encoders, also when they are combined with packetization procedures similar to those proposed by ITU-TG.764, Voice Packetization-Packetized Voice Protocols, or FRF.11, Voice over Frame Relay Implementation Agreement, FR Forum. The result is that there is a low probability of including several packets from different voice sources in the same ATM cell when the number of sources are few, e.g. less than 20. The remaining part of the ATM cell, which is not filled with packets at the expiration of the Timer_CU, is then padded. However, at few sources, the resulting padding overhead very much destroys the advantage of having employed encoders, with or without silence removal. A situation where this could be a problem is when doing PBX trunking. PBX stands for Private Branch Exchange and is essentially an on-premise telephone exchange system that services a number of telephones within a building.
It should therefore be appreciated that there is a need for a device and related system and method for more efficiently transporting packetized data from several sources on a packet oriented transport medium and in particular when there arc only a few sources.
SUMMARY
This invention is embodied in a method according to claims
1
to
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, a related apparatus according to claims
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to
17
and a related system according to claims
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to
According to the invention there is an apparatus and a corresponding method for packetizing at least two data streams, comprising at least two packetizing means, each packetizing means producing first packets from said respective
Jain Raj
Telefonaktiebolaget LM Ericsson (publ)
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