Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2001-07-19
2004-04-20
Dorvil, Richemond (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S500000, C370S477000
Reexamination Certificate
active
06725191
ABSTRACT:
FIELD OF THE INVENTION
The invention relates to transmitting voice over the internet and in particular to reducing bandwidth required to transmit voice over the internet
BACKGROUND OF THE INVENTION
Methods and apparatus for transmitting voice over internet protocol (VOIP) are known. VOIP services are offered by numerous companies and standards for internet telephony have been promulgated by the ITU-T. The ITU-T umbrella standard for VOIP is H.323 rev 2 (1998), “Packet based multimedia communications systems”, the disclosure of which is incorporated herein by reference. An alternative umbrella standard referred to as “Session Initiation Protocol (SIP)” has recently been promulgated for internet telephony by the Internet Engineering Task Force (IETF).
In an internet telephony session between a first and second party, an internet connection is provided between communication equipment at the first party's premises and communication equipment at the second party's premises via their respective internet service providers. During the telephony session each party's communication equipment generates a stream of samples of the party's speech which is parsed into a sequence of groups referred to as “audio frames”. Each audio frame contains a predetermined desired number of samples and corresponds to a desired sampling period. The communication equipment encodes the samples in each audio frame in a constellation of symbols using an appropriate audio encoding scheme such as PCM, ADPCM or LPC.
Each encoded audio frame is encapsulated in a “real time transport packet” in accordance with a real time transport protocol. Under the ITU-T H323 internet telephony standard, the audio frame is encapsulated in an RTP packet in accordance with a real time protocol referred to by the acronym “RTP”. RTP is defined in Schulzrinne, et al., “RTP: A Transport Prototcol for Real-Time Applications”, RFC 1889, Internet Engineering Task Force, January 1996 the disclosure of which is incorporated herein by reference.
In accordance with RFC 1889, the real time transport packet, hereinafter referred to as an “RTP packet”, that encapsulates the audio frame comprises a header having a sequence number. The sequence number corresponds to the temporal order of the audio frame in the RTP packet relative to other audio frames in the sequence of audio frames generated by the communication equipment. Each RTP packet is in turn packaged in a data packet with a suitable data packet header according to an internet transport protocol. Typically, the internet transport protocol for “RTP transmission” is UDP. The data packets are transmitted in a stream of data packets over the internet to the other party.
When the other party receives the stream of data packets, the other party's communication equipment strips each data packet in the stream and its enclosed RTP packet of their respective headers to “unload” the audio frame “payload” in the RTP packet. The communication equipment then concatenates the unloaded audio frames sequentially according to the sequence numbers of their respective RTP packets. The concatenated audio frames are decoded and converted to analogue audio signals to reproduce the speech of the party transmitting the data packets.
Transmission of data packets using UDP can be unreliable and data packets sent via UDP can disappear without a trace and never reach their intended destinations. A data packet can be lost for example if it passes through a network node that is overloaded and “decides” to dump excess traffic. The rate at which data packets are lost generally increases as a network becomes more congested.
To improve reliability and quality of internet telephony using RTP “on top of” UDP and reduce effects of data packet loss on internet telephony, redundancy is sometimes implemented in audio frame transmission between parties to an internet telephony session. With redundancy a same audio frame to be transmitted from one to the other of the parties participating in the internet telephony session is transmitted more than once to assure that it reaches its destination. A redundancy protocol has been promulgated in C. Perkins et al., “RTP Payload for Redundant Audio Data” RTP 2198. Internet Engineering Task Force, September 1997 the disclosure of which is incorporated herein by reference.
While redundancy reduces vulnerability of data transmission to packet loss and improves reliability of data transmission, transmission of data with redundancy generally requires a bit-rate greater than a bit-rate required to transmit the data without redundancy. Redundant data transmission therefore utilizes a greater portion of channel capacity than non-redundant transmission. As a result, while redundancy provides some protection against data packet loss, redundancy tends to increase network congestion, which can in turn exacerbate the packet loss problem redundancy is intended to alleviate. Frugal use of redundancy is therefore generally advisable.
U.S. patent application Ser. No. 09/241,857, entitled “Method & Apparatus for Transmitting Packets”, the disclosure of which is incorporated in its entirety herein by reference, describes a method of implementing redundancy in audio and video data packet transmission over the internet. The method discloses inter alia, controlling use of redundancy in transmitting information over an internet channel responsive to transmission conditions over the channel so as to reduce channel capacity required to support data transmission with redundancy.
SUMMARY OF THE INVENTION
An aspect of some embodiments of the present invention relates to providing a method for transmitting voice over the internet with redundancy that can generally be implemented at average bit-rates that are lower than average bit-rates required by prior art methods of transmitting voice over the internet with redundancy. As a result, a VOIP redundancy method, in accordance with an embodiment of the present invention, generally uses less channel capacity than prior art VOIP redundancy methods.
In accordance with an embodiment of the present invention, the speech of a person participating in an internet telephony session with another person or persons is monitored to determine when the person is speaking and when the person is silent. In addition for periods, hereinafter referred to as “voice periods”, during which the person is speaking, the person's speech is optionally analyzed to determine which portions of the voice periods are stationary.
A stationary portion of a speech period is a time period, having duration equal to duration of at least two audio frames into which a person's speech is parsed for transmission, during which a power spectrum of the voice period is substantially constant. Stationary portions of a voice period are referred to as stationary intervals. Except for a first audio frame that falls entirely within a stationary interval, audio frames that fall entirely within a stationary interval are referred to as stationary audio frames. Audio frames that are not entirely within a stationary interval or audio frames which are a first audio frame completely within a stationary interval are referred to as non-stationary audio frames. By definition, stationary audio frames from a same stationary interval have a same spectrum. As a result any stationary audio frame in a stationary interval can be reconstructed from a previous audio frame in the stationary interval.
In some embodiments of the present invention, as in many prior art VOIP systems, both silent periods and voice periods of the person's speech are encoded in audio frames and transmitted in data packets to the person or persons with whom the person is speaking. However, in accordance with an embodiment of the present invention, if redundancy is required during the telephony session to assure quality of voice transmission, redundancy is implemented only for voice periods of the person's speech and optionally only for non-stationary audio frames of the voice periods. Redundancy is not implemented for th
Dorvil Richemond
Lerner Martin
VocalTec Communications Limited
LandOfFree
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