Remote echo cancellation in a packet based network

Details Remote echo cancellation in a packet based network Remote echo cancellation in a packet based network

2000-05-12

2004-04-20

Nguyen, Chau (Department: 2663)

Multiplex communications

Duplex

Transmit/receive interaction control

Reexamination Certificate

active

06724736

ABSTRACT:

FIELD OF THE INVENTION
The present invention relates generally to voice over IP networks and more particularly relates to an apparatus and method of performing remote echo cancellation for the local endpoint of a connection.
BACKGROUND OF THE INVENTION
Separate Voice and Data Networks
Currently, there is a growing trend to converge voice and data networks so that both utilize the same network infrastructure. The currently available systems that combine voice and data have limited applications and scope. An example is Automatic Call Distribution (ACD), which permits service agents in call centers to access customer files in conjunction with incoming telephone calls. ACD centers, however, remain costly and difficult to deploy, requiring custom systems integration in most cases. Another example is the voice logging/auditing system used by emergency call centers (e.g., 911) and financial institutions. Deployment has been limited due to the limited scalability of the system since voice is on one network and data is on another, both tied together by awkward database linkages.
The aim of IP telephony is to provision voice over IP based networks in both the local area network (LAN) and the wide area network (WAN). Currently, voice and data generally flow over separate networks, the goal is to transmit them both over a single medium and on a single network.
A block diagram illustrating example separate prior art data and voice networks is shown in FIG.
1
. The LAN portion, generally referenced
10
, comprises the LAN cabling infrastructure, routers, switches and gateways
12
and one or more network devices connected to the LAN. Examples of typical network devices include servers
14
, workstations
16
and printers (not shown). The voice portion, generally referenced
20
, has at its core a private branch exchange (PBX)
24
which comprises one or more trunk line interfaces and one or more telephone and/or facsimile extension interfaces. The PBX is connected to the public switched telephone network (PSTN)
22
via one or more trunk lines
28
, e.g., analog T1, E1, T3, ISDN, etc. A plurality of user telephones
26
and one or more facsimile machines
27
are also connected directly to the PBX via phone line extensions
29
.
The paradigm currently in wide spread use consists of circuit switched fabric
20
for voice networks and a completely separate LAN infrastructure
10
for data. Most enterprises today use proprietary PBX equipment for voice traffic.
Voice and Data Over a Shared Network
An increasingly common IP telephony paradigm consists of telephone and data tightly coupled on IP packet based, switched, multimedia networks where voice and data share a common transport mechanism. It is expected that this paradigm will spur the development of a wealth of new applications that take advantage of the simultaneous delivery of voice and data over a single unified fabric.
A block diagram illustrating a voice over an IP network where voice and data share a common infrastructure is shown in FIG.
2
. The IP telephony system, generally referenced
30
, comprises, a LAN infrastructure represented by an Ethernet switch
32
, a router, one or more telephones
36
, workstations
34
, a gateway
42
, a gatekeeper
46
, a PBX
33
with a LAN interface port and a Layer
3
switch
38
. The key components of an IP telephony system
30
are the modified desktop, gatekeeper and gateway entities. For the desktop, users may have an Ethernet phone
36
that plugs into an Ethernet RJ-45 jack or a handset or headset
35
that plugs into a PC
37
.
Today, all LAN based telephony systems need to connect to the PSTN
44
. The gateway is the entity that is specifically designed to convert voice from the IP domain to the PSTN domain. The gatekeeper is primarily the IP telephony equivalent of the PBX in the PSTN world.
Typically, the IP telephony traffic is supported by a packet-based infrastructure such as an Ethernet network but a circuit-based infrastructure can be used as well with some provisions (e.g., ATM LAN emulation on ATM networks). Telephony calls traversing the intranet may pass through a Layer
3
switch
38
or a router (not shown) connecting a corporate intranet
40
. The Layer
3
switch and the router should support Quality of Service (QoS) features such as IEEE 802.1p and 802.1Q and Resource Reservation Protocol (RSVP).
ITU-T Recommendation H.323
The International Telecommunications Union (ITU-T) Telecommunications Standardization Sector has issued a number of standards related to telecommunications. The Series H standards deals with audiovisual and multimedia systems and describes standards for systems and terminal equipment for audiovisual services. The H.323 standard is an umbrella standard that covers various audio and video encoding standards. Related standards include H.225.0 that covers media stream packetization and call signaling protocols and H.245 that covers audio and video capability exchange, management of logical channels and transport of control and indication signals. Details describing these standards can be found in ITU-T Recommendation H.323 (Draft 4 August 1999), ITU-T Recommendation H.225.0 (February 1998) and ITU-T Recommendation H.245 (Jun. 3, 1999).
A block diagram illustrating example prior art H.323 compliant terminal equipment is shown in FIG.
3
. The H.323 terminal
50
comprises a video codec
52
, audio codec
54
, system control
56
and H.225.0 layer
64
. The system control comprises H.245 control
58
, call control
60
and Registration, Admission and Status (RAS) control
62
.
Attached video equipment
66
includes any type of video equipment, such as cameras and monitors including their control and selection, and various video processing equipment. Attached audio equipment
70
includes devices such as those providing voice activation sensing, microphones, loudspeakers, telephone instruments and microphone mixers. Data applications and associated user interfaces
72
such as those that use the T.120 real time audiographics conferencing standard or other data services over the data channel. The attached system control and user interface
74
provides the human user interface for system control. The network interface
68
provides the interface to the IP based network.
The video codec
52
functions to encode video signals from the video source (e.g., video camera) for transmission over the network and to decode the received video data for output to a video display. If a terminal incorporates video communications, it must be capable of encoding and decoding video information in accordance with H.261. A terminal may also optionally support encoding and decoding video in accordance with other recommendations such as H.263.
The audio codec
54
functions to encode audio signals from the audio source (e.g., (microphone) for transmission over the network and to decode the received audio data for output to a loudspeaker. All H.323 audio terminals must be capable of encoding and decoding speech in accordance with G.711 including both A-law and &mgr;-law encoding. Other types of audio that may be supported include G.722, G.723, G.728 and G.729.
The data channel supports telematic application such as electronic whiteboards, still image transfer, file exchange, database access, real time audiographics conferencing (T.120), etc. The system control unit
56
provides services as defined in the H.245 and H.225.0 standards. For example, the system control unit provides signaling for proper operation of the H.323 terminal, call control, capability exchange, signaling of commands and indications and messaging to describe the content of logical channels. The H.225.0 Layer
64
is operative to format the transmitted video, audio, data and control streams into messages for output to the network interface. It also functions to retrieve the received video, audio, data and control steams from messages received from the network interface
68
.
The gateway functions to convert voice from the IP domain to the PSTN domain. In particular, it converts IP packetized voice to a format

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