U.S. patent application number 13/190015 was filed with the patent office on 2011-11-17 for error handling for named signal events in wireless communications.
This patent application is currently assigned to NORTEL NETWORKS LIMITED. Invention is credited to Michael Flynn Thomas.
Application Number | 20110280135 13/190015 |
Document ID | / |
Family ID | 44455454 |
Filed Date | 2011-11-17 |
United States Patent
Application |
20110280135 |
Kind Code |
A1 |
Thomas; Michael Flynn |
November 17, 2011 |
ERROR HANDLING FOR NAMED SIGNAL EVENTS IN WIRELESS
COMMUNICATIONS
Abstract
The present invention provides a technique for handling damaged
named signal event (NSE) packets that are received in association
with a voice over packet based telephony call. The received packets
for a telephony call may include voice packets as well as NSE
packets. Upon receipt of the packets at a physical layer, the
packets are processed at a link layer to detect damaged packets
having link layer errors. The incoming packets are sent to an audio
application at an application layer for further processing. For
damaged packets, link layer error indicia is generated at the link
layer and sent to the audio processing application. The audio
application will use the link layer error indicia to identify
damaged packets or payloads associated with damaged packets, and
determine whether the payloads corresponding to the damaged packets
include NSE information defining an NSE. Damaged NSE packets are
processed accordingly.
Inventors: |
Thomas; Michael Flynn;
(Plano, TX) |
Assignee: |
NORTEL NETWORKS LIMITED
Mississauga
CA
|
Family ID: |
44455454 |
Appl. No.: |
13/190015 |
Filed: |
July 25, 2011 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
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11316428 |
Dec 22, 2005 |
8005029 |
|
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13190015 |
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Current U.S.
Class: |
370/241 ;
370/328 |
Current CPC
Class: |
H04M 7/0084 20130101;
H04M 2207/185 20130101; H04L 1/0061 20130101 |
Class at
Publication: |
370/241 ;
370/328 |
International
Class: |
H04W 4/00 20090101
H04W004/00; H04W 24/00 20090101 H04W024/00 |
Claims
1. A method for handling packets for a telephony call, the method
comprising: receiving packets associated with a telephony call;
detecting damaged packets among the received packets; identifying
damaged named signal event packets among the detected damaged
packets; and providing error processing for the damaged named
signal event packets.
2. The method of claim 1, wherein the received packets comprise
named signal event packets and voice packets, the method further
comprising providing audio processing for the voice packets and for
undamaged named signal event packets.
3. The method of claim 2, wherein the audio processing is provided
for damaged and undamaged voice packets.
4. The method of claim 1, wherein identifying damaged named signal
packets comprises: monitoring a payload of a detected damaged
packet to detect named signal event information defining a named
signal event in the payload.
5. The method of claim 1, wherein providing error processing
comprises discarding a payload of a damaged named signal event
packet.
6. The method of claim 1, wherein the received packets are streamed
using a real time streaming protocol, the received packets
comprising voice packets and named signal event packets.
7. The method of claim 1, wherein a named signal event represents
an in-band signaling.
8. The method of claim 7, wherein the named signal event comprises
at least one tone.
9. The method of claim 8, wherein the named signal event
corresponds to at least one of a ringing signal, a busy signal and
a fast busy signal.
10. The method of claim 9, wherein the named signal event
corresponds to at least one dual-tone multi-frequency tone.
11. An apparatus for handling packets for a telephony call, the
apparatus comprising: a communication interface operable to receive
packets associated with a telephony call; and at least one
processor associated with the communication interface and operable:
to detect damaged packets among the received packets; to identify
damaged named signal event packets among the detected damaged
packets; and to provide error processing for the damaged named
signal event packets.
12. The apparatus of claim 11, wherein the received packets
comprise named signal event packets and voice packets, the at least
one processor being further operable to provide audio processing to
the voice packets and to undamaged named signal event packets.
13. The apparatus of claim 12, wherein the at least one processor
is operable to provide audio processing for damaged and undamaged
voice packets.
14. The apparatus of claim 11, wherein the at least one processor
is operable to identify a damaged named signal packet by:
monitoring a payload of a detected damaged packet to detect named
signal event information defining a named signal event in the
payload.
15. The apparatus of claim 11, wherein the at least one processor
is operable to provide error processing by discarding a payload of
a damaged named signal event packet.
16. The apparatus of claim 11, wherein the received packets are
streamed using a real time streaming protocol, the received packets
comprising voice packets and named signal event packets.
17. The apparatus of claim 11, wherein a named signal event
represents an in-band signaling event.
18. The apparatus of claim 17, wherein the named signal event
comprises at least one tone.
19. The apparatus of claim 18, wherein the named signal event
corresponds to at least one of a ringing signal, a busy signal and
a fast busy signal.
20. The apparatus of claim 19, wherein the named signal event
corresponds to at least one dual-tone multi-frequency tone.
Description
CROSS-REFERENCE TO RELATED APPLICATION
[0001] The present application is a continuation of U.S. patent
application Ser. No. 11/316,428, filed on Dec. 22, 2005, the
disclosure of which is hereby incorporated by reference in its
entirety.
FIELD OF THE INVENTION
[0002] The present invention relates to wireless communications,
and in particular to detecting and processing damaged packets
carrying named signal event information in association with a
telephony call.
BACKGROUND OF THE INVENTION
[0003] The public switched telephone network (PSTN) has
traditionally provided telephony communications to the masses. The
PSTN is a circuit-switched network where each call is essentially
allocated a dedicated circuit through which audio signals are
carried. These audio signals may include voice information as well
as in-band signaling information. The in-band signaling information
is tones or tone sequences for providing audible alerts to one of
the parties to the call as well as for conveying control
information among communication terminals and call processing
entities in the PSTN. For example, the audible alerts may include
the traditional busy, fast busy, and ringing alerts provided to a
caller when the call is being initiated. The control information
may include dual tone multi-frequency (DTMF) tones corresponding to
keys on the telephone terminal's keypad. The DTMF tones may
correspond to dialed digits used to initiate a call or a selection
made by a caller during automated call processing. The events
requiring delivery of in-band signaling information are generally
referred to as named signal events.
[0004] Given the increased capacity and reliability of packet
networks, such as the Internet, telephony communications can now be
supported over the packet networks. Packet-based telephony
communications are often referred to as voice-over-packet (VoP)
communications or, when supported by the Internet Protocol (IP),
voice over IP (VoIP) communications. For VoP telephony, audio
signals are encoded and placed into packets, which are delivered
over the packet networks.
[0005] In an effort to maintain a consistent user experience among
PSTN and VoP telephony, named signal events are used VoP telephony.
However, in VoP telephony the tones associated with the named
signal events are generally not encoded with the voice information.
Instead, a tone description corresponding to the tone is placed in
a packet, which is delivered over the packet network. The tone
description may define the tone or tone sequence as well as length
information describing how long the each tone should be provided
when presented to the intended party or device. Further information
regarding the handling of named signal events for VoP telephony is
provided in Internet Engineering Task Force (IETF) Request For
Comment (RFC) 2833, which is incorporated herein by reference.
[0006] Notably, VoP packets for telephony communications are
streamed from one communication terminal to another over the packet
network in real time. Since there is no time for retransmission of
lost or damaged packets and the nature of voice allows for a
significant number of lost or damaged packets without undue
degradation of the voice signal, error checking is not provided for
received voice packets. When packet networks provide wireless
access for wireless communication terminals, the likelihood of
damaged packets being received increases dramatically over wired
packet networks. The impact of damaged packets on the voice signal
is tolerable; however, the impact on packets carrying named signal
event information is more problematic. Errors may result in
generating the wrong tones, generating tones for the wrong lengths
of time, and the like. Since error checking is often disabled for
VoIP telephony in a wireless environment, there is no way to detect
the errors in the packets carrying named signal event information.
Accordingly, there is need for a way to efficiently and effectively
detect and process errant packets carrying named signal event
information in association with a VoIP telephony call.
SUMMARY OF THE INVENTION
[0007] The present invention provides an efficient and effective
technique for handling damaged named signal event (NSE) packets
that are received in association with a voice over packet based
telephony call. The received packets for a telephony call may
include voice packets as well as NSE packets. Upon receipt of the
packets at a physical layer, the packets are processed at a link
layer to detect damaged packets having link layer errors. Whether
damaged or not, the incoming packets are sent to an audio
application at an application layer for further processing. For
damaged packets, link layer error indication is generated at the
link layer and sent to the audio processing application at the
application layer. The audio application will use the link layer
error indicia to identify damaged packets or payloads associated
with damaged packets, and determine whether the payloads
corresponding to the damaged packets include NSE information
defining a named signal event. If the damaged packets have NSE
information, damaged NSE packets are processed accordingly.
[0008] In one embodiment, damaged NSE packets, and in particular
the payloads corresponding to the damaged NSE packets, are
discarded. Thus, the audio processing application will process
damaged and undamaged voice packets, as well as undamaged NSE
packets, in traditional fashion. The damaged NSE packets are
processed as desired or discarded to minimize the impact of
receiving errant information defining a named signal event. A named
signal event may include any in-band signaling, such as a busy,
fast busy, or ringing tones heard by a caller during initiation of
a call, as well as the delivery of dual tone multi-frequency tones
used for initiating a call or selecting options during automated
call processing.
[0009] Those skilled in the art will appreciate the scope of the
present invention and realize additional aspects thereof after
reading the following detailed description of the preferred
embodiments in association with the accompanying drawing
figures.
BRIEF DESCRIPTION OF THE DRAWING FIGURES
[0010] The accompanying drawing figures incorporated in and forming
a part of this specification illustrate several aspects of the
invention, and together with the description serve to explain the
principles of the invention.
[0011] FIG. 1 is a block representation of a communication
environment according to one embodiment of the present
invention.
[0012] FIG. 2 is a block representation of layer processing
according to one embodiment of the present invention.
[0013] FIG. 3 is a flow diagram illustrating error handling
according to one embodiment of the present invention.
[0014] FIG. 4 is a block representation of an access point
according to one embodiment of the present invention.
[0015] FIG. 5 is a block representation of a mobile terminal
according to one embodiment of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0016] The embodiments set forth below represent the necessary
information to enable those skilled in the art to practice the
invention and illustrate the best mode of practicing the invention.
Upon reading the following description in light of the accompanying
drawing figures, those skilled in the art will understand the
concepts of the invention and will recognize applications of these
concepts not particularly addressed herein. It should be understood
that these concepts and applications fall within the scope of the
disclosure and the accompanying claims.
[0017] Prior to delving into the details of the present invention,
an overview of a wireless communication environment is illustrated
in FIG. 1. As illustrated, a packet network 10 supports telephony
communications with or between any number of mobile terminals 12.
Generally, an access point 16 is used to support wireless
communications with the mobile terminals 12 and will directly or
indirectly connect to the packet network 10. For local wireless
communications, such as those facilitated using Bluetooth or
wireless local area network (WLAN) standards, such as those set
forth in the IEEE's 802.11 standards, the access point 16 will
provide a relatively limited local wireless communication zone in
which local wireless communications are supported, and will connect
to the packet network 10 through an access network 14. In a
cellular environment, the access point 16 may take the form of a
cellular base station that connects to a cellular network 18 via a
mobile switching center (MSC) 20. The cellular network 18 will
directly or indirectly connect to the packet network 10.
[0018] The present invention can be implemented in either the
access point 16 or the mobile terminal 12, and functions to detect
and process damaged packets that carry named signal event (NSE)
information in association with a telephony call. The packets
carrying NSE information will be referred to as NSE packets. The
NSE packets are provided in the same data stream as the streaming
voice packets, if the telephony call has been established and voice
information is present. Accordingly, during a telephony call, a
stream of packets flowing between the access point 16 and a mobile
terminal 12 may include NSE packets intermingled with the voice
packets. During initiation of the telephony call or during silent
periods during the telephony call, the NSE packets may be
intermingled with packets presenting silence, or may simply be
delivered as necessary.
[0019] Instead of allowing errant NSE packets to be delivered and
processed along with other packets associated with a telephony
call, the present invention detects damaged NSE packets at a link
layer and alerts an audio application that processes the incoming
voice signals for the telephony call. The audio application can
then process the damaged NSE packets as necessary. In one
embodiment, the damaged NSE packets are discarded and NSE packets
arriving undamaged are processed in normal fashion.
[0020] With reference to FIG. 2, a block representation of the
basic operation of the present invention is illustrated with
respect to a wireless communication protocol model. Although
additional layers may be employed, the model illustrated provides
five layers: a physical layer (Layer 1), a link layer (Layer 2), a
network layer (Layer 3), a transport layer (Layer 4), and an
application layer (Layer 5+). The physical layer (Layer 1) provides
the hardware mechanisms responsible for sending and receiving
wireless signals. The link layer (Layer 2) affords transmission
protocol knowledge and management, as well as handling errors in
the physical layer, flow control, and frame or packet
synchronization. The link layer is often divided into two logical
sub-layers: the media access control (MAC) layer, and a logical
link control (LLC) layer. The MAC sub-layer controls network access
transmission permission. The LLC sub-layer controls error checking,
flow control, and frame synchronization. The network layer (Layer
3) provides routing or forwarding functions, perhaps based on the
Internet Protocol (IP).
[0021] The transport layer (Layer 4) operates to provide
transparent transfer of voice information between communication
endpoints, such as the mobile terminals 12 and the communication
terminal for the remote party engaged in the telephony call. The
transport layer is responsible for controlling packet flow from
endpoint to endpoint, and may operate to ensure complete transfer
of the information from one endpoint to another. The transport
layer for VoP telephony may use the Real-Time Protocol (RTP) to
control the streaming of voice packets and any NSE packets
associated with a telephony call. The User Datagram Protocol (UDP)
is employed in the transport layer for RTP communications to
facilitate delivery of the RTP information being communicated from
one endpoint to another. Using the UDP checksum, the receiver can
detect damage in packets. For a VoIP wireless call, error detection
will typically be disabled. Thus, the transport layer cannot detect
damaged packets for a VoP telephony call, and the information
carried in the damaged packets will be provided to the application
layer (Layer 5+) for normal audio processing by an audio processing
application and would result in improper behavior.
[0022] For the present invention, an error checking function 22 is
provided at the link layer (Layer 2), wherein received NSE packets
from the physical layer (Layer 1) are monitored to detect errors
occurring at the physical layer. These errors are essentially radio
frequency (RF) errors recognizable at the link layer. The error
checking function 22 will monitor all incoming packets and detect
damaged packets. Upon detecting a damaged packet, the link layer
error indicia for the damaged packet is provided to the audio
processing application at the application layer (Layer 5+). The
audio processing application will correlate the link layer error
indicia with a corresponding packet or packet payload, and
determine if the packet is an NSE packet by looking for NSE
information in the payload of the packet. If the payload includes
NSE information, and thus the packet is an NSE packet, the packet
may be dropped or processed as desired.
[0023] Turning now to FIG. 3, a flow diagram is provided to
illustrate receiving streaming packets associated with a telephony
call, and in particular, processing the NSE packets when damage is
detected. Initially, the transport layer (Layer 4) error checking
is turned off or otherwise deactivated (step 100). At some point,
packets associated with a telephony call are received at the
physical layer (Layer 1) (step 102) and processed at the link layer
(Layer 2) (step 104). The packets may be NSE packets, voice
packets, or a combination thereof. The received packets are
monitored for link layer errors with the error checking function 22
(step 106). For each packet, if there is no link layer error (step
108), the packet is provided to the application layer (Layer 5+)
where the packet is processed in normal fashion by an audio
application or the like (step 110).
[0024] If a link layer error is detected for a given packet by the
error checking function 22 (step 108), error indicia is created and
passed to the application layer (Layer 5+) indicating that a link
layer error was detected for the packet (step 112). The audio
application at the application layer will receive the link layer
error indicia and determine if the damaged packet was an NSE packet
by seeing if the payload contains named signal event information
(step 114). If the packet was an NSE packet (step 116), the payload
of the NSE packet is discarded at the application layer (step 118)
or processed as desired. If the packet is not an NSE packet (step
116), the packet payload is processed in normal fashion at the
application layer by the audio application (step 110). Accordingly,
the audio application can receive error indicia from the link layer
indicating that the packet is damaged. If the damaged packet is an
NSE packet, the payload for the damaged NSE packet can be dropped
to minimize the impact of the damaged packet on the voice
processing, and thus the telephony call.
[0025] With reference to FIG. 4, an access point 16 configured
according to one embodiment of the present invention is
illustrated. The access point 16 generally includes a control
system 24, a baseband processor 26 providing the audio applications
28 and error checking function 22, transmit circuitry 30, receive
circuitry 32, an antenna 34, and a network interface 36, which may
connect to the MSC 20 in a cellular embodiment or an access network
14 in a WLAN embodiment. The receive circuitry 32 receives radio
frequency signals through the antenna 34 bearing packets associated
with telephony calls from mobile terminals 12. Preferably, a low
noise amplifier and a filter (not shown) cooperate to amplify and
remove broadband interference from the signal for processing.
Downconversion and digitization circuitry (not shown) will then
downconvert the filtered, received signal to an intermediate or
baseband frequency signal, which is then digitized into one or more
digital streams.
[0026] The baseband processor 22 processes the digitized received
signal to extract the information or data bits conveyed in the
received signal. Again, the error checking function 22 and the
audio applications 28 may be implemented in the baseband processor
26. This processing typically comprises demodulation, decoding,
error detection, and audio processing. As such, the baseband
processor 26 is generally implemented in one or more digital signal
processors (DSPs). The received information is then sent across a
wireless network via the network interface 36 or transmitted to
another mobile terminal 12 serviced by the access point 16.
[0027] On the transmit side, the baseband processor 26 receives
digitized data, which may represent voice, data, or NSE
information, from the network interface 36 under the control of
control system 24, and encodes the data for transmission. The
encoded data is output to the transmit circuitry 30, where it is
modulated by a carrier signal having a desired transmit frequency
or frequencies. A power amplifier (not shown) will amplify the
modulated carrier signal to a level appropriate for transmission,
and deliver the modulated carrier signal to the antenna 34.
[0028] With reference to FIG. 5, a mobile terminal 12 configured
according to one embodiment of the present invention is
illustrated. Similarly to the access point 16, the mobile terminal
12 will include a control system 38, a baseband processor 40
providing the audio applications 28 and error checking function 22,
transmit circuitry 42, receive circuitry 44, an antenna 46, and
user interface circuitry 48. The receive circuitry 44 receives
radio frequency signals through the antenna 46 bearing information
from one or more access points 16. Preferably, a low noise
amplifier and a filter (not shown) cooperate to amplify and remove
broadband interference from the signal for processing.
Downconversion and digitization circuitry (not shown) will then
downconvert the filtered, received signal to an intermediate or
baseband frequency signal, which is then digitized into one or more
digital streams. The baseband processor 40 processes the digitized
received signal to extract the information or data bits conveyed in
the received signal. This processing typically comprises
demodulation, decoding, error correction, and audio processing.
[0029] For transmission, the baseband processor 40 receives
digitized data, which may represent voice, data, or NSE
information, from the control system 38, which it encodes for
transmission. The encoded data is output to the transmit circuitry
42, where it is used by a modulator to modulate a carrier signal
that is at a desired transmit frequency or frequencies. A power
amplifier (not shown) will amplify the modulated carrier signal to
a level appropriate for transmission, and deliver the modulated
carrier signal to the antenna 46.
[0030] Those skilled in the art will recognize improvements and
modifications to the preferred embodiments of the present
invention. All such improvements and modifications are considered
within the scope of the concepts disclosed herein and the claims
that follow.
* * * * *