Voice gateway and route selection

Details Voice gateway and route selection Voice gateway and route selection

1999-08-16

2003-06-24

Vanderpuye, Ken (Department: 2661)

Multiplex communications

Pathfinding or routing

Switching a message which includes an address header

C370S352000

Reexamination Certificate

active

06584110

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a technique of transmitting voice signals through an IP (internet protocol) network by selecting a route according to an allowable delay time.
2. Description of the Related Art
Conventional voice transmission using a circuit switching network guarantees a bandwidth and a delay time. Voice transmission using an IP network, however, does not guarantee a bandwidth or a maximum delay time, and therefore, causes a problem of the quality of transmitted voice. It is known in the IP network voice transmission that an available bandwidth and a delay time vary depending on the delays in routers and gateways for relaying IP packets and on the congestion in transmission lines. Unlike general data transmission, voice or video transmission allows only a short delay time because voice or video data requires real-time transmission.
FIG. 1
shows static route selection in an internet telephone system. The system includes a gateway
10
serving the caller side to transmit voice, gateway
20
serving the receiver side to receive the voice, a circuit switching network
30
, and route selection data
11
used by the gateway
10
on the caller side. A transmission terminal (not shown) makes a call. The call is received by the gateway
10
, which relays the call to one of the gateways
20
. When relaying the call to a destination telephone, the gateway
10
selects a route according to an IP address related to a destination phone number. At this time, the gateway
10
refers to the route selection data
11
that contains some routes provided with priority. If a route of the first priority is unavailable, the gateway
10
selects a route of the second priority, and so on. For example, the gateway
10
first selects a route (a) of the first priority from the route selection data
11
and relays the call to the gateway
20
as indicated with {circumflex over (1)}. If a return message from the gateway
20
, passed as indicated by {circumflex over (2)}, indicates that a bandwidth is unavailable, the gateway
10
selects a route (b) of the second priority and relays the call to another gateway
20
as indicated with {circumflex over (3)}. If a return message from the gateway
20
, passed as indicated by {circumflex over (4)}, indicates that a bandwidth is available, a call connection process will be completed.
Thereafter, the gateways
10
and
20
exchange voice compression rules available for the IP address related to the call, and the gateway
20
selects one of the rules. The selected rule is used between the gateway
10
on the caller side and the gateway
20
on the receiver side to compress and decompress voice data.
This prior art fixes IP addresses and routes for phone numbers in advance, and therefore, is unable to dynamically change routes from one to another. Namely, the prior art is unable to determine a route depending on a change in a delay time in an IP network
50
, which will be explained later with reference to FIG.
2
.
FIG. 2
shows dynamic route selection in an internet telephone system.
A gatekeeper
40
is provided for gateways
10
and
20
. The gateways
10
and
20
periodically inform the gatekeeper
40
of bandwidth data. Upon receiving a call, the gateway
10
refers to the gatekeeper
40
as well as route selection data
11
and determines a route that involves an available bandwidth. Namely, the gateway
10
selects a route depending on dynamic changes in the conditions of the IP network
50
.
The gateways
10
and
20
have each an address conversion function and a voice compression-decompression function. The address conversion function is used to relate a phone number in a circuit switching network
30
to an IP address in the IP network
50
. The voice compression-decompression function effectively uses the bandwidths of the IP network according to voice compression rules shown. in Table 1.
TABLE 1
Voice compression rules (excerpts from ITU
standards)
Bit rate
5/6
8
16
48/56/64
>128
Bandwidth
Kbps
Kbps
Kbps
Kbps
Kbps
4 KHz
Standard
G.723.1
G.729
G.728
G.711
(Tel.
quality)
Comp.
30 ms
10 ms
5 ms
0.75 ms
time
7 KHz
Standard
G.722
15 KHz
Standard
MPEG
1/2
Table 1 shows voice compression rules (part) standardized by ITU-T (International Telecommunication Union Telecommunication Standardization Sector). There are voice signal bandwidths of 5/6 Kbps, 8 Kbps, 16 Kbps, 48/56/64 Kbps, and 128 Kbps or over. A compression time becomes longer as the voice signal bandwidth becomes lower. Namely, if a long delay time is allowed, a route that involves a compression rule for a high compression rate can be selected.
FIG. 3
shows voice compression rules according to an H.323 protocol of ITU-T.
For a better understanding of the present invention, the H.323 protocol will be roughly explained. The details of the H.323 protocol are described in the eighth section “Call Signaling Procedures” in pages 45 to 78 of ITU-T H.323 recommendation, September 1997.
The H.323 protocol is a title standard including a plurality of protocols as shown in FIG.
3
. These protocols will be explained.
G.711 and G.723.1 are voice encoding protocols.
G.711 relates to PCM (pulse code modulation) and samples voice at 8 KHz to form encoded data of 64 Kbps.
G.723.1 forms encoded data of 5.3.Kbps (ACELP) or 6.3 Kbps (MP-MLQ).
H.261 is a protocol for encoding video data for a videoconference. There are CIF (common interface format) (288×352) and QCIF (quarter CIF).
H.225.0(RTP) (real time protocol) forms packets from voice and video streams and carries out synchronization based on time stamps.
H.225.0(RTCP) (real time control protocol) controls the RTP.
H.225.0(RAS) is a signal protocol between a terminal and a gatekeeper. According to this signal protocol, the gatekeeper certifies a connection request from the terminal.
H.225.0(Q.931) is a call control signal protocol based on Q.931.
H.245 transfers control signals between terminals. The control signals represent, for example, the performance of the terminals.
Each message has a logic channel and is described in ASN.1 syntax.
FIG. 4
shows basic H.323 protocol phases. Phase A is a call setup, phase B carries out initial communication and capability exchange, phase C establishes audiovisual communication, phase D is a call service, and phase E is a call termination.
a) Phase A: Setup
FIG. 5
shows call control messages based on H.225.0 exchanged between end points
1
and
2
in the call setup of the phase A. The end point
1
transmits a message with an IP address to the end point
2
. The end point
2
returns a call proc. message, an alert message, or a connect message each containing an H.245 control channel address used for H.245 signaling to the end point
1
. Then, the phase A ends.
b) Phase B: Initial Communication and Capability
The phase B sets an H.245 control channel. The H.245 control channel is used to open a media channel and exchange transmission capabilities between the end points
1
and
2
. As an option, the end point
2
on the receiver side may set an H.245 control channel when receiving a setup message, or the end point
1
on the caller side may set an H.245 control channel when receiving the alerting or call proc. message. Basic messages will be explained.
An H.245 TerminalCapabilitySet message is used to exchange capabilities between terminals.
An H.245 MasterSlaveDetermination message is used to determine a master between two terminals according to a random number.
According to the H.323 protocol, a caller terminal sends a TerminalCapabilitySet message to a receiver terminal to inform the receiver terminal of a terminal capability. If the terminal capability is acceptable, the receiver terminal returns a TerminalCapabilityAck message to the caller terminal. The exchange of terminal capabilities may be carried out any time. For this purpose, a TCP channel for an H.245 control channel must be set in advance.
c) Phase C: Establishment of Audiovisual Communication
The phase C sets logic channels for various pieces of data after the transmission capa

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