US 7203164 B2
A control terminal such as a CMTS is initialized to receive packets of voice calls having parameters including a bit rate, a packetization interval, and a call identification. A plurality of queues is created to define a corresponding plurality of phases at a sub-multiple of the packetization interval. Voice calls are admitted to the control terminal. The voice calls are distributed among the queues in a predetermined order as the voice calls are admitted and the voice calls removed from the queues as the voice calls are terminated. USGs are periodically issued at the phases defined by the queues. The USGs include a call identification and a grant of bandwidth sufficient to transmit the packets.
1. A method for timing the transmission of unsolicited grants (USGs) of bandwidth to transmit voice packets on a shared transmission medium comprising the steps of:
initializing a control terminal to receive packets of voice calls having parameters including a bit rate, a packetization interval, and a call identification;
creating a plurality of queues corresponding to a plurality of phases defined at a sub-multiple of the packetization interval;
admitting voice calls to the control terminal;
distributing the voice calls among the queues in a predetermined order as the voice calls are admitted;
removing the voice calls from the queues as the voice calls are terminated; and
periodically issuing at the phases corresponding to the queues USGs that include a call identification and a grant of bandwidth sufficient to transmit the packets.
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This application claims priority of the following provisional patent applications: Application No. 60/182,470, filed on Feb. 15, 2000; Application No. 60/247,188, filed on Nov. 9, 2000; Application No. 60/254,415, filed on Dec. 8, 2000; Application No. 60/262,203, filed on Jan. 17, 2001; Application No. 60/262,201, filed on Jan. 17, 2001. This application is also a continuation-in-part of pending non-provisional application Ser. No. 09/427,792, filed on Oct. 27, 1999. The disclosures of these applications are incorporated herein fully by reference.
This invention relates to a voice architecture for transmission over a shared, contention based medium, and more particularly, to transmission of multiple voice calls on such a medium.
Traditionally voice is supported in telephone networks as a circuit switched service with a dedicated slot assignment. As a result, resources are dedicated for the entire duration of the call, which results in under-utilization of the bandwidth. However, a such service results in little jitter and minimal delay.
More recently, voice is being supported over internet protocol (IP), shipped as IP frames over data networks. This results in higher utilization of bandwidth at the cost of delay and jitter. There are many standards that digitize analog voice in trade offs of efficiency, bandwidth, processing power, and voice quality. Voice quality manifests itself in low jitter and small delay. Users of voice transmission facilities pay according to the Quality of Service (QoS).
According to the invention, voice calls that occur at a packetization interval are divided into phases within the packetization interval for call management. This limits or “bounds” the jitter that inherently occurs transmission of packetized voice to the phase interval, instead of the packetization interval. Specifically, a control terminal such as a cable modem termination system or a telephone central office is initialized to receive packets of voice calls having parameters including a bit rate, a packetization interval, and a call identification. A plurality of queues is created to define a corresponding plurality of phases at a sub-multiple of the packetization interval. Voice calls are admitted to the control terminal. The voice calls are distributed among the queues in a predetermined order as the voice calls are admitted and the voice calls removed from the queues as the voice calls are terminated. In a cable transmission system, unsolicited grants (USGs) arc periodically issued at the phases defined by the queues. The USGs include a call identification and a grant of bandwidth sufficient to transmit the packets.
The features of specific embodiments of the best mode contemplated of carrying out the invention are illustrated in the drawings, in which:
In CM 13, a block 14 represents an upper layer such as a web browser or other source of data to be transmitted. The data is fed to block 14 from input queues 15, which store the received data to be transmitted upstream until it is processed by CM 13. Packet data units (PDUs) from upper layer 14 are coupled to output queues 16. Output queues 16 send to a cable modem scheduler (CMSC) 18 an indication of the queue state so CMSC 18 can piggyback a request in a region of data minislots 11 a. CMSC 18 sends requests for minislots to a burst multiplexer 20, which forms the physical layer interface between CM 13 and upstream channel 11. CMSC 18 also transmits send indications to output queues 16 to transfer PDUs to burst multiplexer 20 at the appropriate time to fill the assigned time slots. Responsive to the requests from CMSC 18, burst multiplexer 20 sends requests for minislots to burst multiplexer 20 for transmission in contention minislots 11 b and PDUs to burst multiplexer 20 for transmission in minislots 11 a assigned as described below by grants from CMTS 12.
In CMTS 12, bursts of data are coupled from upstream channel 11 to a burst demodulator 22, which forms the physical layer interface between CMTS 12 and upstream channel 11. Burst demodulator 22 directs requests for minislots to a request queue 24 and PDUs to a block 26, which represents an upper layer such as a web browser or other data receiver. PDUs from upper layer 26 are sent to output queues 28 for external use. A contention slot allocator (CSA) 30 specifies which time slots are in the upstream channel are to be used as assigned data minislots 11 a and contention minislots 11 b. If there are no requests in queue 24, contention minislots 11 b are assigned. The piggyback (PB) probability is sent from request queue 24 to CSA 30 to help CSA 30 determine the breakdown of upstream bandwidth into data minislots 11 a and contention minislots 11 b. The state of queues 24 is sent to call admission controller (CAC) 32, where voice calls are processed and a call admission signal is generated if the call is accepted. The call admission signal is sent in downstream channel 10 to the CM that is requesting a call admission. An upstream scheduler (USCH) 34 is connected to CAC 32 to control the management of voice calls. As discussed in more detail below, parameters for setting up the queues in USCH 34 are sent thereto by CAC 32 and USCH 34 acknowledges that the queues are set up. USCH 34 receives information from CSA 30 representative of the distribution of contention minislots (CMS). A block 36 monitors contention minislots 11 b on upstream channel 11 on a continuous basis to detect a collision immediately if a collision occurs. In the event of a collision, block 36 sends a contention/no contention (C/NC) signal to a block 38, which performs a collision resolution algorithm (CRA). Block 36 sends a collision message (C/NC) on downstream channel 10 to enable the CM to resend the collided request. USCH 34 computes the minislots to be granted to the CM based on the requests received on upstream channel 11 and the CMS region, which are sent to a Map builder 40. The resulting MAPS generated by Map builder 40 are introduced into the MPEG transport stream of downstream channel 10 as control messages.
At CM 13, the MAP messages are recovered with the other control messages by a MPEG demultiplexer 42. The grants in the MAP messages and slot structure messages for the particular CM are separated and sent to CMSC 18 to control the allocation of data to the minislots 11 a in upstream channel 11. The slot structure messages specify whether the respective slots are contention minislots or data minislots and whether the minislots are for the particular CM or another CM. The collision (C/NC) signal transmitted on downstream channel 10 and a CRA parameter derived from a control message are sent to a downstream CRA 44 for processing. Downstream CRA 44 sends the count of contention minislots Lo CM scheduler 18.
Reference is made to
An early indication of a call going silent is desirable so the CMTS reduces the grant size without delay and to minimize jitter. A codec is incorporated into each CM to convert the voice calls to binary voice packets. (Either one codec could operate on all the calls on a time shared basis or a separate codec could be provided for each call.) In either case, as shown in
This voice jitter effect can be reduced by subdividing the upstream frames into a number of smaller phases and employing a separate queue for each phase. The jitter value can be bounded by selection of the phase size. At the time of making a call request, the SID of the cable modem can be used to specify the phase size and thus the maximum jitter value. As illustrated in a diagram (c), the 10 ms frames are subdivided into four 2.5 ms phases, namely Phases 1 to 4. Each of these phases is managed by a queue having a 16 call capacity. Assuming that the frame carries 40 calls, these calls are distributed so the queue of Phase 1 is full with 16 calls, the queue of Phase 2 is full with 16 calls, the queue of Phase 3 is half full with 8 calls, and the queue of Phase 4 is empty. Filling one queue to capacity before distributing calls to the next phase, as illustrated in diagram (c), makes it possible to use the remaining bandwidth in a frame for data transmission without unduly fragmenting the date. Calls do not cross phase boundaries when they are re-ordered. In contrast with diagram (b), if all the calls, except call C40, are terminated, for example, the location of call C40 changes by only four positions, as illustrated in a diagram (d), namely, to the edge of phase 3, instead of by 39 positions to the edge of the frame, as illustrated in diagram (b). In the worst case, a call would change by 15 positions, instead of 63 positions by virtue of the subdivision of the frames into phases, each of which is supported by a separate queue.
In the case of diagram (b), the maximum jitter that could be experienced by a call is 10 ms. In the case of diagram (d), the maximum jitter that could be experienced by a call is reduced to 2.5 ms. When a new call is requested, the maximum jitter could be specified by configuring the queues to limit the phases as illustrated.
Jitter can also be caused by silence suppression. If a voice stream is disabled by switch 64 (
Given this periodicity, this time stamp can be suppressed at the CM and generated at the CMTS, thereby eliminating the RTP header in each voice packet. This amounts to 12 bytes of savings per voice packet. In summary, the VoIP headers are suppressed and replaced by a voice packet header comprising a silence bit 74 and a voice identification field 76. The voice transmission burst comprise a silence bit 74, a voice id field 76, and a voice payload 78. After transmission, the suppressed VoIP headers are expanded. The suppressed headers could be stored in a table and recovered from the table when expanded.
It should be noted that calls at a 5 ms interval are distributed to all the queues, calls at a 10 ms interval are distributed to every other queue, and calls at a 20 ms interval are distributed to one queue. Calls are distributed to fill the earlier phases first. Thus, in general one phase is fuller than the preceding phase, e.g., Phase 1 is filler than Phase 2 and Phase 2 is fuller than Phase 3. As these queues are filled a counter is decremented to reflect the state of the queue.
With reference to
It should be noted that the calls are distributed so the state of the queues is as balanced, i.e., as nearly the same, as possible. Referring to
The described invention could be implemented in software or hardware. A software implementation may be somewhat easier to design and cheaper to build in small quantities. Although a hardware implementation may be more difficult to design, it may operate faster and the cost becomes less significant in large quantities. The described embodiment of the invention is only considered to be preferred and illustrative of the inventive concept; the scope of the invention is not to be restricted to such embodiment. Various and numerous other arrangements may be devised by one skilled in the art without departing from the spirit and scope of this invention. In general, as illustrated in
This application claims priority to the following provisional applications:
U.S. Patent Ser. No. 60/182,470, entitled “intelligent Silence Suppression,” filed Feb. 15, 2000, by Gummalla et al., (still pending) (incorporated by reference in its entirety herein).
U.S. Patent Ser. No. 60/247,188, entitled “A Local Scheduling Mechanism for Cable Modems,” filed Nov. 9, 2000, by Sala et al., (still pending) (incorporated by reference in its entirety herein).
U.S. Patent Ser. No. 60/254,415, entitled “A Local Scheduling Mechanism for Cable Modems,” filed Dec. 8, 2000, by Sala et al., (still pending) (incorporated by reference in its entirety herein).
U.S. Patent Ser. No. 60/262,201, entitled “Voice Scheduling Algorithms,” filed Jan. 17, 2001, by Sala et al., (still pending) (incorporated by reference in its entirety herein).
U.S. Patent Ser. No. 60/262,203, entitled “Concatenation of Requests at CMTS,” filed Jan. 17, 2001, by Sala et al., (still pending) (incorporated by reference in its entirety herein).
This application claims priority to the following non-provisional application:
U.S. patent Ser. No. 09/427,792, entitled “System and Method for Multiplexing Data from Multiple Sources,” filed Oct. 27, 1999, by Limb et al., (still pending) (incorporated by reference in its entirety herein).
This application is related to the following non-provisional applications, all having the same filing date as the present application:
“Method, System and Computer Program Product for Scheduling Upstream Communications”, U.S. patent Ser. No. 09/783,404 by Gummalla et al. (incorporated by reference in its entirety herein).
“System and Method for Suppressing Silence in Voice Traffic over an Asynchronous Communication Medium,” U.S. patent Ser. No. 09/783,405 by Gummalla et al., (incorporated by reference in its entirety herein).
“System and Method for Combining Requests for Data Bandwidth by a Data Provider for Transmission of Data Over an Asynchronous Communication Medium,” U.S. patent Ser. No. 09/783,311 by Gummalla et al., (incorporated by reference in its entirety herein).
“Cable Modem System and Method for Specialized Data Transfer,” U.S. patent Ser. No. 09/783,403 by Bunn et al., (incorporated by reference in its entirety herein).