Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching
Reexamination Certificate
1998-02-04
2001-07-31
Marcelo, Melvin (Department: 2739)
Multiplex communications
Pathfinding or routing
Combined circuit switching and packet switching
C370S503000
Reexamination Certificate
active
06269095
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to computer telephony and, in particular, to a Voice over IP (VoIP) application.
2. Description of the Related Art
The use of the Internet for real-time voice applications is increasingly widespread. A key to real-time speech and voice transmission on the Internet is the Voice over IP (VoIP) gateway. The Voice over IP gateway bridges the public switched telephone network (PSTN) or integrated services digital network (ISDN) with the packet-switched data network (TCP/IP Local Area Network). Such a VoIP gateway is configured to provide IP call control and IP data transport, which includes compression/decompression of voice channels using G.723.1 vocoding. In addition, PSTN or ISDN call control and compression and packetization are provided, typically using G.711 vocoding.
An exemplary gateway system
1000
is illustrated in FIG.
1
. The gateway system
1000
includes an IP Network Interface
1002
, a host computer
1006
, a voice payload data processing unit
1008
, and a PBX interface
1004
. The IP Network Interface
1002
is coupled, for example, to a 10 MBit Ethernet. The PBX Interface
1004
is coupled, for example, to a plurality of E1/TI/PRI lines. The host computer
1006
includes a conversion unit
1006
a
and a fax processing unit
1006
b,
and provides overall system control.
Voice payload processing is accomplished at the Voice Payload Data Processing Unit
1008
. The voice payload data processing unit
1008
may include a plurality of digital signal processors (DSP). Typically, one DSP handles the call processing (e.g., real-time vocoding, silence suppression, echo cancellation, DTMF filtering, and &mgr;-law/a-law conversion) of three or four channels. The IP Network Interface
1002
performs IP Network packetizing for received voice payload data packets from the voice payload data packet unit
1008
. This includes, for example, encapsulating the data using RTP, UTP, IP and Ethernet headers. The gateway system
1000
may support both voice and fax operations.
A key to successful VoIP gateway functionality is the minimization of encoding/decoding delays at the DSPs. Delay causes two problems; echo and talker overlap. Delay may result from the need to collect a frame of voice samples to be processed by the voice coder, the process of encoding and collecting the encoded samples into a packet for transmission over the packet network, or network delays which are caused by the physical medium and protocol used to transmit the voice data and by the buffers used to remove packet jitter on the received side. As discussed above, in a typical VoIP gateway, a three or four telephony channels are compressed/decompressed by a single DSP which then provides the data to the IP Network Interface.
FIG. 2
illustrates a single DSP. For sake of clarity, the host computer
1006
and additional DSPs are omitted. Gateway
100
includes a digital signal processor (DSP)
102
, a network interface
104
, as well as a telephone interface
106
. As shown, a G.723.1 encoding/decoding module
108
is provided for converting the received pulse code modulation (PCM) signals into signals suitable for IP transmission. The telephony interface
106
interfaces with the telephone network. As shown, three 64 kilobit per second telephony channels
110
,
112
and
114
are fed into the DSP
102
. The channels
110
,
112
and
114
are not synchronized. Accordingly, multiplexing delays at the DSP
102
may be introduced which can have a detrimental effect upon system performance. The multiplexing delay can be anywhere between zero milliseconds and thirty milliseconds, minus processing time for encoding/decoding.
The multiplexing delays are illustrated in greater detail in FIG.
2
. In particular, the telephony interface
106
separates the received ISDN data stream into three (for example) B-channels, designated channel A, channel B, and channel C, respectively. (Typically, the telephony data is provided as a constant stream, which is buffered, processed, buffered, and sent out). As shown, voice channels A, B and C are received approximately simultaneously according to a single clock. Thus, one 30-millisecond frame may be ready for encoding for each incoming telephony channel at the same time, designated T
0
. Since the channels are being received simultaneously, a packet for each channel A, B, and C is also available (simultaneously) at times T
1
−T
4
. As can be seen, the G.723.1 encoding of channel A, represented by reference A
1
, is completed at time T
1
+&Dgr;T. The encoding of channel B, represented by reference B
1
, is not completed until T
1
+2&Dgr;T; and the encoding of channel C, represented by reference C
1
, is not completed until T
1
+3&Dgr;T. A similar result is obtained for the encoding of subsequent voice packets. As can be seen, there may be a large multiplexing delay on channel C. As shown, a delay of 3&Dgr;T is required before channel C is encoded.
Accordingly, there is a need for an improved voice-over IP gateway which minimizes system delays. In particular, there is a need for a VoIP gateway which minimizes multiplexing delays at the DSP.
SUMMARY OF THE INVENTION
These and other problems in the prior art are overcome in large part by a VoIP gateway according to the present invention. According to an embodiment of the present invention, incoming telephony channels are synchronized prior to transmission to the vocoder to minimize multiplexing delays. In particular, in one embodiment employing three 64 kilobit per second telephony channels with 30 millisecond frames, a 10 millisecond delay is introduced between each channel. In this fashion, system delays may be minimized. For example, the channels may be separately initialized at 10 millisecond intervals.
According to one embodiment of the invention, a gateway is provided including a telephony interface, a network interface, and a digital signal processor (DSP) for encoding the incoming telephony channels. The telephony interface is configured to separate an incoming data stream into separate channels. The telephony interface is further configured to control operation of a plurality of clocks for synchronizing the transfer of the channels to the encoding DSP.
REFERENCES:
patent: 4903261 (1990-02-01), Baran et al.
patent: 5442630 (1995-08-01), Gagliardi et al.
patent: 5479407 (1995-12-01), Ko et al.
patent: 5526353 (1996-06-01), Henley
patent: 5680552 (1997-10-01), Netravali et al.
patent: 97 23078 (1997-06-01), None
patent: 97 47118 (1997-12-01), None
“A Packet Telephony Gateway for Public Network Operators”, Houghton TF et al. ; Iss '97 World Telecommunications Congress, Global Network Evolution: Convergence or Collision? Toronto, Sep. 21-26, 1997, vol. 2, pp. 35-44.
“Real-Time Voice Over Packet-Switched Networks”;Kostas TJ et al.;, IEEE Network: The Magazine of Computer Communications, vol. 12, No. 1, , Jan. 1, 1998, pp. 18-27.
Dumas Gregory
Neubauer Harald
Skrzynski Mark
Marcelo Melvin
Siemens Information and Communication Networks Inc.
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