Multiplex communications – Data flow congestion prevention or control – Flow control of data transmission through a network
Reexamination Certificate
1998-02-20
2001-04-17
Chin, Wellington (Department: 2664)
Multiplex communications
Data flow congestion prevention or control
Flow control of data transmission through a network
C370S395430, C370S389000
Reexamination Certificate
active
06219339
ABSTRACT:
FIELD OF THE INVENTION
This invention relates generally to communications and, more particularly, to packet voice systems.
BACKGROUND OF THE INVENTION
In packet voice systems, as in any packet-based system, periods of traffic congestion may result. This traffic congestion adds delay to the packet system and, as such, impacts the ability of people to effectively carry on a conversation. Congestion also results in packet losses due to buffer overflow, and creates degradation in quality of voice communication.
One approach to alleviating traffic congestion is to selectively drop entire voice packets. Generally, the selectively dropped voice packets represent “less-important” speech such as “semi-silence” (e.g., see Nanying Yin, San-Qi Li, and Thomas E. Stern, “Congestion Control for Packet Voice by Selective Packet Discarding,” IEEE Trans. on Communications, May 1990, Vol. 38, No. 5; Kotikalapudi Sriram, R. Scott McKinney, and Mostafa Hashem Sherif, “Voice Packetization and Compression in Broadband ATM Networks,” IEEE Journal on Selected Areas in Communications, Vol. 9, No. 3, April 1991; and David W. Petr, Luiz A DaSilva, Jr., Victor S. Frost, “Priority Discarding of Speech in Integrated Packet Networks,” IEEE Journal on Selected Areas in Communications, Vol. 7, No. 5, June 1989). (For completeness only, it should be noted that there are also other alternatives such as ITU (International Telecommunications Union) Embedded Adaptive Differential Pulse Code Modulation (ADPCM) Standard G.727, which describes a method for dropping the least significant bits of a voice sample within a packet.)
When entire voice packets are dropped, subjective voice quality is affected. For example, it is well known that subjective voice quality degradation is virtually imperceptible for a random packet loss rate of up to 1 in 100 for wavefrom encoded speech such as pulse code modulation (CM) and ADPCM (e.g., see J. G. Gruber and N. Le, “Performance requirements for integrated voice/data networks,” IEEE Journal on Selected Areas in Communications, December 1983, pp. 981-1005; and N. S. Jayant and S. W. Christensen, “Effects of packet losses in waveform coded speech and improvements due to an odd-even sample interpolation,” IEEE Trans. on Communications, February 1981, pp. 101-109).
Unfortunately, selective discarding of entire voice packets as a function of the type of speech, e.g., semi-silence, may cause consecutive packets from the same voice source to be discarded. This phenomenon is shown in
FIG. 1
, which illustrates the nature of the packet voice arrival process (from multiple sources). During periods of congestion, the packets from a few sources tend to arrive at periodic intervals at the tail ends of bursts of packets from all sources. This is shown in
FIG. 1
by burst
1
, burst
2
, and burst
3
, and circles
11
,
12
, and
13
. Consequently, in these periods these few sources experience excessive loss of consecutive packets due to buffer overflow, while packets of other sources do not experience such excessive loss. The loss of consecutive packets further degrades subjective voice quality and may become noticeable to the receiving party. As such, to avoid the perceptive quality degradation due to consecutive packet loss, the packet system is often designed for a much lower packet loss rate (e.g., about 1 in 1000 (or less) in the case of ADPCM encoded speech) for the voice quality degradation due to consecutive packet losses to remain imperceptible to the listener. Such a low packet loss rate requirement, of course, results in operating the system at much lower engineered traffic load and hence higher cost of the packet system.
SUMMARY OF THE INVENTION
We have discovered an alternative approach to selectively discarding packets. In particular, when a transmission buffer reaches a predefined level, discarding of a packet is performed as a function of previously discarded packets.
The inventive concept is applicable to packet systems in general, e.g., Asynchronous Transfer Mode (ATM), Internet Protocol (IP), Frame Relay (ITU-T G.764 packet format) systems, etc. As such, one embodiment is a generic form of packet system. In this system, a transmission buffer stores voice packets for transmission, each voice packet comprising a sequence number, the values of which range from 0 to n−1, and a source identifier, k. When traffic congestion is detected, the transmitter portion of the system selectively discards one packet from a source k at the output of the transmission buffer if no packet from source k was dropped in either the last n−1 packets from source k or over a predefined prior interval of time.
In another embodiment of the invention, a packet voice system includes an AAL2 and Service Specific Convergence Sublayer (SSCS) System. In this system, a transmission buffer stores AAL2 voice packets for transmission, each AAL2 voice packet comprising a sequence number, the values of which range from 0 to n−1, and a source identifier, k. When traffic congestion is detected, the transmitter portion of the SSCS System selectively discards one packet from a source k at the output of the transmission buffer if no packet from source k was dropped in either the last n−1 packets from source k or over a predefined prior interval of time.
In another embodiment of the invention, a packet voice system includes an AAL2 and Service Specific Convergence Sublayer (SSCS) System. In this system, a transmission buffer stores AAL2 voice packets for transmission, each AAL2 voice packet comprising a sequence number, the values of which range from 0 to n−1, and a source identifier, k. When traffic congestion is detected, the transmitter portion of the SSCS System selectively discards one packet from a source k at the input of the transmission buffer if no packet from source k was dropped in either the last n−1 packets from source k or over a predefined prior interval of time.
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John G. Gruber and Nguyen H. Le, “Performance Requireiments for Integrated Voice/Data Networks”, IEEE Journal on Selected Areas in Communications, vol. SAC-I, No. 6, Dec. 1983.
Nuggehally S. Jayant and Susan W. Christensen, “Effects of Packet Losses in Waveform Coded Speech and Improvements Due to an Odd-Even Sample-Interpolation Procedure”, IEEE Transactions on Communicatinos, vol. COM-29, No. 2, Feb. 1981.
David W. Petr, Luiz A. DaSilva, Jr., and Victor S. Frost, “Priority Discarding of Speech in Integrated Packet Networks”, IEEE Journal on Selected Areas in Communications, vol. 7, No. 5, Jun. 1989.
Kotikalapudi Sriram, R. Scott McKinney, and Mostafa Hashem Sherif, “Voice Packetization and Compression in Broadband ATM Networks”, IEEE Journal on Selected Areas in Communications, vol., 9, No. 3, Apr. 1991.
Nanying Yin, San-Qi Li, Thomas E. Stern, “Congestion Control for Packet Voice by Selective Packet Discarding”, IEEE Transactions on Communications, vol. 38, No. 5, May 1990.
Doshi Bharat Tarachand
Sriram Kotikalapudi
Wang Yung-Terng
Chin Wellington
Lucent Technologies - Inc.
Opalach Joseph J.
Pham Brenda H.
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