WO2011099840A2

WO2011099840A2 - A centralized network system with codec selection scheme

Info

Publication number: WO2011099840A2
Application number: PCT/MY2010/000260
Authority: WO
Inventors: Rajina M A Raj Mohamed; Khong Neng Choong; Mazlan Abbas; Azizul Rahman Sharif; Osman Mohd Said
Original assignee: Mimos Berhad
Priority date: 2010-02-09
Filing date: 2010-11-10
Publication date: 2011-08-18
Also published as: WO2011099840A3; MY153840A

Abstract

A centralized VoIP network system (10) comprises a plurality of endpoints (12a-12f) and a SIP server (11) coupled to said endpoints (12a-12f) in said network (10), said server (11) operable to determine the codec bitrate from the network bandwidth availability based on the predetermined threshold values.

Description

A Centralized Network System with Codec Selection Scheme

Field of Invention

The present invention relates to a network system, more particularly relates to a centralized codec selection in VoIP network system.

Background of Invention

Packet-based networks such as Voice over Internet Protocol (VoIP) packets networks, are rapidly emerging as a viable alternative to traditional switched network systems. This is because VoIP can be used for voice transport in that it allows live voice conversations to be integrated with existing IP data and image applications. VoIP systems use a codec during a call to convert analog signals to a compressed digital bit stream. The codecs allow VoIP systems to transmit encoded audio, video or data across Internet Protocol networks. Each codec has an associated bandwidth, which is a factor in determining the bandwidth allocated for_. a call along with other factors such as the bandwidth required for the network protocol headers and the packetization rate. A codec with a higher bandwidth provides a better voice quality. However, VoIP packets are dela sensitive and traffic congestion could easily affect th performance of a VoIP session.

Hence, VoIP application should be enhanced with the capability of switching codec where each codec operates at different bitrate as to stay functional at different traffic conditions. A codec should be adjustable during voice communication according to the network availability.

In conventional method, codec switching or selection is done at endpoint's device or terminal. This causes an unfair codec selection among various endpoints during high network utilization. As a consequence, some endpoints might experience bad audio quality as higher codec is being utilized by other endpoints. Also, battery lifetime is shortened because endpoint's terminal might continuously search for an optimum codec. Figure 1 depicts a diagram showing a network system with the conventional method. All endpoints get high bit rate codec as all endpoints prefer the best codec for a better audio communication which is considered an unfair codec selection among various endpoints. The network bandwidth is not able to support many endpoints as high bit rate codec consume more bandwidth and some endpoints even has no opportunity to get connected. In this case, the codec selection is based on First-Come-First-Serve basis (FCFS) .

Therefore, there is a need of the present invention to provide a system having codec selection which is decided by a server based on bandwidth availability, where all endpoints will potentially experience optimum audio quality and at the same time, able to conserve battery energy. The codec selection is evenly distributed (fair) among endpoints which certain endpoints might get a high bit rate and the rest get the middle and so forth based on current bandwidth availability. With the present invention, more endpoints can be supported by this system.

Other objects of this invention will become apparent on the reading of this entire disclosure.

Summary of Invention

In one aspect of the present invention, a centralized VoIP network system comprises a plurality of endpoints and a SIP server coupled to the endpoints, the server operable to determine the codec bitrate from the network bandwidth availability based on the predetermined threshold values. In a case where the available network bandwidth exceeds a first predetermined threshold value during the call initiation, a high bitrate codec will be chosen. In a case where the available network bandwidth is lower than a second predetermined threshold value during the call initiation, a low bitrate codec will be chosen. And in a case where the available network bandwidth is between the first and second predetermined threshold values during the call initiation, a medium bitrate codec will be chosen.

A method of operating a centralized VoIP network system, the method includes receiving an invite (which includes an SDP payload) at a SIP server from a first endpoint to initiate a call session to a second endpoint, determining codec bitrate at the SIP server from the network bandwidth availability based on the predetermined threshold values to be sent to the second endpoint, accepting the invitation by the second endpoint by sending a message with the codec selection information to the server which then forward to the first endpoint to establish the call session. Brief Description of the Drawings

Other objects, features, and advantages of the invention will be apparent from the following description when read with reference to the accompanying drawings. In the drawings, wherein like reference numerals denote corresponding parts throughout the several views:

Figure 1 depicts a diagram showing a network system with the conventional method;

Figure 2 illustrates the structure of the centralized voice over internet protocol (VoIP) network system of the present invention;

Figure 3 shows a sequence diagram showing an operation of the network system of the present invention; and

Figure 4 is a flow chart showing an operation of the network system of the present invention. Detailed Description of the Preferred Embodiments

In the following detailed description, numerous specific details are set forth in order to provide a thorough understanding of the invention. However, it will be understood by those of ordinary skill in the art that the invention may be practiced without these specific details. In other instances, well-known methods, procedures and/or components have not been described in detail so as not to obscure the invention. Reference will now be made in detail to the preferred embodiments of the present invention, examples of which are illustrated in the accompanying drawings.

Referring to Figure 2 , a centralized codec network system of the present invention generally designated as numeral reference 10. This system (10) using a method to select a suitable codec for endpoints (12) which is determined by a server (11) according to the bandwidth availability. Figure 2 illustrates the structure of the voice over internet protocol (VoIP) network system (10) in an exemplary embodiment of the invention. The network system (10) includes a plurality of endpoints (12a-12f) and at least one server (11). The endpoints (12a-12f) and server (11) are connected to a network (14) such as the Internet. The server (11) is a session initiation protocol (SIP) server which manages endpoints (12a-12f) in the network system (10) that register with it and provides call processing functionality. All endpoints (12a-12f) and potentially more are connected to the wireless access point and registered to the server (11) to communicate to another party. Endpoints (12a-12f) may be any devices that implement functionality for initiating and terminating calls such as computers, laptops, PDA, VoIP phones, etc .

Figure 2 shows a VoIP communication session between two endpoints (12a, 12b) is established through the network (10) where first endpoint (12a) registered to the server (11) to communicate with the second endpoint (12b). The server (11) will then check the network bandwidth availability to provide an optimum audio quality to all endpoints in the same network and send back a notification to the first endpoint (12a) with a suitable codec rate. Then the audio session between the first (12a) and second endpoint (12b) is established with the codec rate suggested by the server (11) and the connection for the current traffic network is preserved. Figure 3 is a sequence diagram and figure 4 is a flow chart showing an operation of the network system (10) of the present invention. For example, the first endpoint (12a) invites (31) the second endpoint (12b) via the server (11) to start a VoIP session by sending an INVITE (SDP) (21) from the first endpoint (12a) to the server (11) as shown in Figures 3 and 4. The server (11) then checks (32) the network bandwidth availability and set the best codec based on the current network scenario to be sent together with the message INVITE (SDP) (22) to the second endpoint (12b) .

The network bandwidth availability is based on two bandwidth thresholds which is configurable by the service provider. In this case, we assumed TH1=80% and above represents a good bandwidth capacity and TH2=50% and below represents a bad bandwidth capacity. Therefore, if the available bandwidth is larger than TH1 (33) , the high bit rate codec will be chosen (34) else if network bandwidth is below than TH2 (35), the server (11) will recommend the low bit rate codec (36) . And when the network bandwidth is between TH1 and TH2 (37), then the medium bit rate codec is chosen (38) for usage. If network bandwidth is not sufficient to support any call (39), then the call is rejected (40) . The second endpoint (12b) accepts the invitation and responds thereto by sending back 200 OK (23) to the server (11) . The server (11) then sends the 200 OK message (24) together with the codec selection information to the first endpoint (12a) . Upon the second endpoint (12b) receiving ACK (25) transmitted from the first endpoint (12a), the RTP audio session (26) is established.

As will be readily apparent to those skilled in the art, the present invention may easily be produced in other specific forms without departing from its essential characteristics. The present embodiments is, therefore, to be considered as merely illustrative and not restrictive, the scope of the invention being indicated by the claims rather than the foregoing description, and all changes which come within therefore intended to be embraced therein .

Claims

1. A centralized VoIP network system (10) comprising:

a plurality of endpoints (12a-12f) ; and

a SIP server (11) coupled to said endpoints (12a-12f) in said network (10), said server (11) operable to determine the codec bitrate from the network bandwidth availability based on the predetermined threshold values.

2. The centralized VoIP network system (10) as claimed in claim 1, wherein said in a case where the available network bandwidth exceeds a first predetermined threshold value during the call initiation, a high bitrate codec will be chosen.

3. The centralized VoIP network system (10) as claimed in claim 1, wherein said in a case where the available network bandwidth is lower than a second predetermined threshold value during the call initiation, a low bitrate codec will be chosen.

4. The centralized VoIP network system (10) as claimed in claim 1, wherein said in a case where the available network bandwidth is between the first and second predetermined threshold values during the call initiation, a medium bitrate codec will be chosen.

5. The centralized VoIP network system (10) as claimed in claim 1, wherein said codec bitrate is sent together with the invitation message by a first endpoint (12a) to a second endpoint (12b) .

6. A method of operating a centralized VoIP network system (10), said method comprising the steps of:

receiving an invite (SDP) (21) at a SIP server (11) from a first endpoint (12a) to initiate a call session to a second endpoint (12b) ;

determining codec bitrate at said SIP server (11) from the network bandwidth availability based on the predetermined threshold values to be sent to said second endpoint (12b) ;

accepting said invitation by said second endpoint (12b) by sending message to said server (11); and

sending said message with the codec selection information by said server (11) to said first endpoint to establish said call session.

7. The method as claimed in claim 6, wherein said in a case where the available network bandwidth exceeds a first predetermined threshold value during the call initiation, said method further comprising the step of choosing a high bitrate codec by said server (11) to be sent to said second endpoint (12b).

8. The method as claimed in claim 6, wherein said in a case where the available network bandwidth is lower than a second predetermined threshold value during the call initiation, said method further comprising the step of choosing a low bitrate codec by said server (11) to be sent to said second endpoint (12b) .

9. The method as claimed in claim 6, wherein said in a case where the available network bandwidth is between the first and second predetermined threshold values during the call initiation, said method further comprising the step of choosing a low bitrate codec by said server (11) to be sent to said second endpoint (12b) .

10. The method as claimed in claim 6, wherein said in a case where the available network bandwidth is not sufficient to support any call (39) , said method further comprising the step of terminating the call by said server (11) .