WO2008040109A1

WO2008040109A1 - Voice and data communication devices, methodologies and systems

Info

Publication number: WO2008040109A1
Application number: PCT/CA2006/001636
Authority: WO
Inventors: Michael Raz; James Mcavoy; Paul De Grandpre
Original assignee: PROLITY Corp
Current assignee: PROLITY Corp
Priority date: 2006-10-04
Filing date: 2006-10-04
Publication date: 2008-04-10
Anticipated expiration: 2009-04-04
Also published as: CA2662529A1

Abstract

A method for half-duplex communication between a first Session Initiation Protocol (SIP) Communication Device having access to a computer network, and a second SIP Communication Device having access to that computer network. The method comprises the steps of establishing a first Real-Time Transport Protocol (RTP) channel between the first SIP Communication Device and a first computer on the network, and establishing a second RTP channel between the second SIP Communication Device and a second computer on the network, then establishing a communication channel between the first computer and the second computer. Transmission access on the commumcaion channel between the first and second computers is then limited to a maximum of one computer at a time.

Description

METHODOLOGIES AND SYSTEMS

FIELD OF THE INVENTION

The present invention relates to data communication and telephony, and more particularly, relates to voice and data equipment, systems and methodologies for use on wired or wireless networks.

BACKGROUND OF THE INVENTION

Internet Protocol based telephony and data equipment, systems and methodologies have become widely adopted as a relatively inexpensive manner for transmitting and receiving voice and other data communications. In this domain, Session Initiation Protocol (hereinafter "SIP") has been widely adopted as a protocol which was developed for the purposes of providing a standardized protocol for initiating, modifying and terminating video, voice, instant messaging or other communication and is one of the leading signal protocols for Voice Over IP (hereinafter "VOIP"). SIP is used primarily for the purposes of establishing and disconnecting voice or video calls on the Internet or in a similar environment. In the context of a typical session, the SIP is utilized to establish and disconnect the call, while the Real-Time Transport Protocol (hereinafter "RTP") is utilized to carry the actual voice, video or data content. SIP and RTP are both Internet Engineering Task Force protocols.

SIP telephones and other SIP communication devices (all of which are generally referred to in the industry as "User Agents" and are referred to herein as either a "SIP telephone", "SIP Communication Device" or "User Agent") are generally known and may be used in the context of Vo [P systems for basic point-to-point telephone calls utilizing full duplex channels. When used in the context of a telephone, the SIP telephone digitizes voice data, and using the SIP and RTP protocols, transmits a voice call or other data transmission, by means of the Internet or other similar wired or wireless environment to other SIP or VoIP capable telephone users. A variety of different systems employ SIP technology, which may be implemented in a variety of different ways. For example, SIP telephones may have the SIP technology incorporated therein in example, a personal computer or other device. In some embodiments, the SIP telephones may communicate directly with one another over the Internet or other network, or alternatively may communicate, by means of one or more servers, with another SIP telephone.

Disadvantageous^, SIP telephones do not presently work in a half duplex environment, and do not presently provide one to many (broadcast) capabilities.

It is desirable that a SIP telephone be capable of application were the network channel is half- duplex, or where one-to-many (broadcast) communication is desired. It is also desirable that a SIP telephone may be used in a PUSH TO TALK (also known as "PRESS TO TRANSMIT" and hereinafter referred to as "PTT") environment (PTT referring to the communication devices such as two- way radios and walkie-talkies, where the user PUSHES a "TALK" or "TRANSMIT" button to initiate the TALK or TRANSMIT mode and whereupon the user gains control of the floor or channel being utilized, permitting the user to send out a communication, while in the meantime, and as long as the user has activated the TALK or TRANSMIT button, preventing other users of the floor or channel from gaining control of the floor or channel being utilized). When the TALK or TRANSMIT button is not pressed or activated, the device is in receive data (or voice) mode and will provide the user with any transmissions that are received on the selected channel or frequency.

SUMMARY OF THE INVENTION

Accordingly, one object of the present invention is to provide an application which permits one or more SIP telephones or other SIP devices to be connected by way of the Internet or other network providing a simulated PUSH TO TALK type environment for the users of thereof.

Another object of the present invention is to provide an application which may permit full duplex telephony equipment such as a SIP telephone or other device to function in a half duplex mode and/or a half duplex environment. i o r telephone or other device to broadcast voice or other data to multiple users on the Internet or other network.

According to one aspect of the present invention, there is provided a method for half-duplex communication between a first SIP Communication Device having access to a computer network and a second SIP Communication Device having access to that computer network, comprising the steps of establishing a first RTP channel between the first SIP Communication Device and a first computer on the network establishing a second RTP channel between the second SIP Communication Device and a second computer on the network establishing a communication channel between the first computer and the second computer limiting transmission access on the communication channel between the first computer and the second computer to a maximum of one computer at a time.

BRIEF DESCRIPTION OF THE DRAWINGS

A preferred embodiment of the present invention is described below with reference to the accompanying drawings, in which:

Figure 1 is an illustration of an analogy between a pair of SIP telephones interacting with the PTT Proxy Server application of one embodiment of the present invention which has been installed on two processors, and a reduced traffic flow pattern at a construction site;

Figure 2 is an illustration of the SIP telephone initiating a session with the PTT Proxy Server application in one embodiment of the present invention;

Figure 3 is an illustration of the establishment of an RTP channel between the PTT Proxy Server application and the SIP telephone in one embodiment of the present invention;

Figure 4 is an illustration of a request being made by a first user of a SIP telephone to gain exclusive access to the channel in one embodiment of the present invention; the first user to access the network in half-duplex mode, making it available to other PTT Proxy Server applications on the network and to other users, in one embodiment of the present invention;

Figure 6 is an illustration of the first user releasing the floor in one embodiment of the present invention;

Figure 7 is an illustration of the relationship between the PPT Proxy Server application of one embodiment of the present invention, the RTP stacks and buffers created thereby, and a SIP telephone and network.

DESCRIPTION OF THE PREFERRED EMBODIMENT

In a preferred embodiment, an application is provided which is referred to herein as a PTT Proxy Server Application (sometimes hereinafter referred to as the "PXS Application") which may be installed on one or more computers, processors or other devices which have access to the Internet or other network system. In the preferred embodiment, the PXS Application when accessed by, for example, a SIP telephone, provides the user with a VoIP environment through which to communicate with other SIP telephones, or alternatively, if selected by the user, provides the user with PTT functionality within the VoIP environment. That is, the PXS Application allows a SIP telephone to operate in its conventional mode, or allows it to function similarly to a PTT device, and to broadcast audio or other data to all phones or devices connected to a wired or wireless network (hereinafter "network"). In one embodiment of the present invention, both the SIP telephone (in software form) and the PXS Application (in software form) may be installed on the same computer, the SIP telephone application presenting to the user an interface which provides the user with the functionality of a SIP telephone. In an alternative embodiment of the present invention, the PXS Application may be installed on a separate server remote from the SIP telephone device or computer. Also in this embodiment, it may be that the other SIP telephones are either hardware devices, or software based SIP telephone devices installed on a computer or other processor. Application installed on a device or computer is somewhat analogous to an individual 2 at the end of a construction site 4 on a two lane road 12 which has required the two lane road 12 to be reduced to single lane traffic 6, which individual together with their counterpart 8 at the other end of the construction site 4 (analogous to a second computer or device on which the PXS Application is installed), controlling access to the single usable lane 6 using "Stop/Go" signs 10. Both individuals coordinate traffic flow in either direction along the single usable lane 6, by giving traffic in one direction access to the single usable lane 6 while preventing traffic in the opposite direction from accessing that lane, and thereafter making the single usable lane 6 available to traffic in the opposite direction while preventing traffic in the first direction from accessing that lane. In this way, the two individuals give access to the lane or prevent access to the lane to ensure that at no time is traffic passing through the lane in both directions.

On a system utilizing the PXS Application installed on two or more computers or other devices on a network to permit half duplex communication between a first SIP telephone 20 and a second SIP telephone 22 over a channel 16 on the network, the PXS Application serves an analogous function to those two individuals at the construction site, first by negotiating which SIP telephone has exclusive access to transmit on the half duplex channel 16 on the network (analogous to the single lane) and then, when the half duplex channel 16 on the network is no longer being transmitted on by that SIP telephone, communicating to the other PXS Applications installed on other computers or devices on the network that the half duplex channel 16 on the network is free to be transmitted on.

In the preferred embodiment, when the PXS Application is installed on a computer or other device, as illustrated in Figure 2, the SIP telephone may request the initiation of a session by sending a SIP session initiation request ("INVITE" 24) to the PXS Application to start a session, whereupon the PXS Application will send a response to the SIP telephone ("OK" 26) in the appropriate manner that the SIP protocol dictates that the SIP telephone is to move to the "connected state". The movement by the SIP telephone to the connected state is acknowledged by the SIP telephone transmitting to the PXS Application the appropriate SIP protocol dictated acknowledgment ("ACK" 28). Through this process, as illustrated in Figure 3, an RTP session other data may be transmitted between the SIP telephone 10 and the PXS Application, which voice or other data will not be transferred to the Internet or other network, until the SIP telephone obtains "the floor" as more fully described herein).

As illustrated in Figure 4, by activating the PUSH button 32 on the SIP telephone 20, a first user is requesting "the floor" (ie. exclusive access to transmit on the channel), which request is transmitted 34 from the SIP telephone to the PXS Application, which PXS Application thereafter and in response thereto sends a Real-time Transport Control Protocol ("RTCP") 36 packet to all other PXS Applications installed on the network, communicating to them that a user now has "the floor" (in the preferred embodiment, this is done by the PXS application transmitting an RTCP APP packet to all other PXS Applications installed on the network), and to temporarily restrict or "lock out" transmission access on the channel, whereupon the other PXS Applications installed on other devices on the network temporarily restrict or "lock out" transmission access to the channel by that device as more fully described herein. It is understood that there are a variety of different ways of providing this information to the other devices on which the PXS Application is installed which are known to persons skilled in the art.

As illustrated in Figure 5, as the first user, while activating the PUSH button 32 now has exclusive access to transmit on the channel, the PXS Application allows RTP voice/audio or data packets 38 to be transmitted or broadcast through the network to other hosts upon which the PXS Application is installed, allowing the transmitted voice/audio or other data packets 38 to be available to, for example, other SIP telephones 22 or devices.

As illustrated in Figure 6, by de-activating the PUSH button 32 on the SIP telephone, the first user releases "the floor" (ie. exclusive access to transmit on the channel), which request is transmitted 40 from the SIP telephone 20 to the PXS Application 14, which thereafter sends a Real-time Transport Control Protocol ("RTCP") signal 42 (an RTCP APP packet again being utilized for the transmission of this information to the other devices on the network on which the PXS Application is installed) to all hosts on the network on which the PXS Application is installed, informing them that "the floor" is now free to be used for transmission by other users network.

Referring to Figure 7, in one embodiment of the invention, the PXS Application installed on a processor with access to RAM or other suitable and accessible memory, establishes a first "ingress" RTP stack 60 and buffers 64A, 64B, 64C and 64D (as more fully described herein) within the device upon which the PXS Application has been installed.

In the case of the ingress RTP stack 60, the PXS Application establishes an RTP Receiver Buffer 64A (for temporarily storing voice or other data received from the SIP telephone 78), an RTP Transceiver Buffer 64B (for temporarily storing voice or other data to be transmitted to the SIP telephone 78), an RTCP Receiver Buffer 64C (which in the preferred embodiment, is not utilized, it being understood that its creation is a matter of convenience and formality, and in compliance with the RTP Protocol, but not strictly necessary in the context of implementing the present invention), and an RTCP Transceiver Buffer 64D (which in the preferred embodiment, is not utilized, it being understood that its creation is a matter of convenience and formality, and in compliance with the RTP protocol, its creation not strictly being necessary in the context of implementing the present invention).

The "ingress" RTP stack 60 has four possible states:

1. "Inactive", that is, no session has been established with a SIP telephone;

2. "Send Only", that is, the "ingress" RTP stack 60 and buffers will only send RTP packets to the SIP telephone;

3. "Receive Only" that is, the "ingress" RTP stack 60 and buffers will only receive RTP packets from the SIP telephone;

4. "Send and Receive" that is, the "ingress" RTP stack 60 and buffers can send and receive RTP packets to/from the SIP telephone.

In the preferred embodiment, the "ingress" RTP stack 60 may be either in the "Inactive" state or the "Send and Receive" state. In an alternative embodiment, the other states may be utilized as would be understood by a person skilled in the art. (represented by lines 61 A and 61B) between the SIP telephone 78 and the first "ingress" RTP stack 60 and buffers 64A and 64B, which first "ingress" RTP stack 60 and buffers 64A and 64B receive communication from (as illustrated by the line 61 A), and direct communication to (as illus1τated by the line 61B), the SIP telephone 78. The ingress stack 60, at all times while the session with the SIP telephone is active, remains in the "Send and Receive" state allowing full duplex communication between the SIP telephone 78 and the device on which the PXS application is installed.

The PXS Application also establishes a second "egress" RTP stack 62 and buffers 68A, 68B, 68C and 68D (as more fully described herein) within the device upon which the PXS Application has been installed.

hi the case of the egress RTP stack 62, the PXS Application establishes an RTP Transceiver Buffer 68A (for temporarily storing voice or other data to be transmitted to the network 80), an RTP Receiver Buffer 68B (for temporarily storing voice or other data received from the network 80), an RTCP Transceiver Buffer 68C (for temporarily storing RTCP data to be transmitted to the network 80), and an RTCP Receiver Buffer 68D (for temporarily storing RTCP data are received from the network 80).

The "egress" RTP stack 62 has four possible states:

1. "Inactive", that is, no session has been established with the network;

2. "Send Only", that is, the "egress" RTP stack 62 and buffers will only send RTP and RTCP packets to the network (and all other devices on the network on which the PXS Application has been installed will be prevented from sending RTP and RTCP packets to the network, as more fully described herein);

3. "Receive Only", that is, the "egress" RTP stack 62 and buffers will only receive RTP and RTCP packets from the network (and any other device on the network on which the PXS Application has been installed will be permitted to send RTP and RTCP packets to the network, as more fully described herein);

4. "Send and Receive" that is, the "egress" RTP stack 62 and buffers can send and receive In the preferred embodiment, the "egress" RTP stack 62 may be either in the "Inactive" state or the "Send Only" state or the "Receive Only" state. In an alternative embodiment, the "Send and Receive" state may be utilized to achieve full duplex communications as would be understood by a person skilled in the art.

The PXS Application also establishes a second session (represented by lines 63A, 63B, 63C and 63D) between the second "egress" RTP stack 62 and buffers 68A, 68B, 68C and 68D and the network, the second "egress" RTP stack 62 creates, maintains and controls a session between the PXS Application and the Internet or other network 80. The egress RTP stack 62, upon the initiation of a session with the network, defaults to a "Receive Only" state.

A first de-activatable communication link 66 is provided by the PXS Application between the RTP Receiver Buffer 64A and the RTP Transceiver Buffer 68 A (the link capable of being activated (that is, being permitted to pass voice or other data therethrough) or deactivated (that is, being temporarily prevented from passing voice or other data therethrough) by the PXS application code). It is understood that in one embodiment, the communication link 66 is activated when the PXS application permits the passage of voice or other data therethrough and is deactivated when the PXS application does not permit the passage of voice or other data therethrough. In its default state, the first de-activatable communication link 66 is "deactivated".

A second communication link 67, which in the preferred embodiment is activated at all times while there is an active session in place with the SIP telephone 78, is also provided by the PXS Application between the RTP Transceiver Buffer 64B and the RTP Receiver Buffer 68B.

In the preferred embodiment, when the user of the SIP telephone 78 has not activated the Push to Talk button, the egress stack 62 is in the "Receive Only" state (the default state), the PXS application will maintain the communication link 66 between the RTP Receiver Buffer 64A and the RTP Transceiver Buffer 68 A in a "de-activated" mode preventing the transfer of voice or other data from the RTP Receiver Buffer 64A to the RTP Transceiver Buffer 68 A, thereby . _ from the RTP Receiver Buffer 64A to the RTP Transceiver Buffer 68A and thereafter to the Internet or other network 80 (when the RTP Receiver Buffer 64 A is filled, if voice/data continues to arrive from the first SIP telephone 78, until such time as the PUSH to TALK button is activated, the buffer will continue to be overwritten with the new voice/data).

When the user of the SIP telephone 78 activates the Push To Talk Signal as described above, the PXS application on the device associated with that SIP telephone, on receipt of that signal, changes the egress stack 62 to a "Send Only" state, and transmits (by way of the RTCP Transceiver Buffer 68C) to all other devices on the network on which the PXS Application is installed, an RTCP APP Packet, signaling to the other devices on the network on which the PXS Application is installed that a user now has "the floor". Additionally, the PXS Application on the first device "activates" the communication link 66 between the RTP Receiver Buffer 64A and the RTP Transceiver Buffer 68 A permitting voice or other data to be transferred therebetween, and to thereby permit the transfer of voice or other data from the SIP telephone 78 through the RTP Receiver Buffer 64A and the RTP Transceiver Buffer 68A and to the Internet or other network 80.

When the user "deactivates" the Push to Talk button, as described above, the PXS application on the first device transmits (by way of the RTCP Transceiver Buffer 68C) to all other devices on the network on which the PXS Application is installed, an RTCP APP Packet informing them that the user now has released "the floor" and changes the egress stack 62 back to a "Receive Only'^' state. Additionally, the PXS Application on the first device "deactivates" the communication link 66 between the RTP Receiver Buffer 64A and the RTP Transceiver Buffer 68A, preventing voice or other data to be transferred therebetween, and to thereby prevent the transfer of voice or other data from the SIP telephone 78 through the RTP Receiver Buffer 64 A and the RTP Transceiver Buffer 68A and to the Internet or other network 80.

When "the floor" is accessible for transmission, another SIP telephone on the network may gain exclusive access to the floor in an analogous manner, by activating the Push to Talk button on that other SIP telephone (the PXS Application on the processor associated with that other SIP manner as described above), which activation of the PUSH TO TALK button will direct the PXS Application associated with that other SIP telephone to transmit an RTCP APP Packet to all devices on the network on which the PXS Application is installed, informing them that another user now has exclusive access to "the floor", which RTCP APP Packet is received by the RTCP Receiver Buffer 68D in recipient processors on which the PXS Application is installed, whereupon the PXS Application in the recipient processors temporarily "lock out" the possible activation of the first communication link 66 associated with their processors, permitting the transmission of voice or other data from the RTP Receiver Buffer 68B to the RTP Transceiver Buffer 64B, thereby allowing the transmission of voice or other data from the network 80, to the SIP telephone 78, by way of the RTP Receiver Buffer 68B to the RTP Transceiver Buffer 64B. Similarly, when the second SIP telephone deactivates the Push to Talk button on that SIP telephone (returning the second SIP telephone to the default "Receive Only" state in the manner described above), which directs the PXS Application associated with that second SIP telephone to transmit an RTCP APP Packet to all devices on the network on which the PXS Application is installed, informing them that another user now has released "the floor", which RTCP APP Packet is received by the RTCP Receiver Buffer 68D on the recipient processors on which the PXS Application is installed. The PXS application on the recipient processor also "unlocks" the previously "locked out" first communication link 66 on the recipient processor (which remains in the deactivated state).

While this invention has been described in the context of providing PTT functionality to a SIP telephone, it is understood that this invention has much broader application. For example, this invention may also be utilized in the context of simulating a wargame scenario, in which the SIP telephone's PUSH TO TALK and broadcast characteristics provide a familiar working environment similar to walkie-talkies, two way radios and other PUSH TO TALK devices. Additionally, it allows a SIP telephone to be utilized both in the context of normal, full-duplex operations (in which case, the PXS Application behaves transparently to the user, allowing the SIP telephone call to proceed over a full-duplex channel in the typical manner) and where a half- duplex environment is to be utilized, the PXS Application allows the SIP telephone to function in a half duplex manner. thereon, a button, or button combination may be selected from the SIP telephone's button set to function as the "PUSH" button to permit all SIP telephones, whether or not they have a unique or preset "PUSH" button thereon, to function in the half-duplex or PUSH TO TALK mode.

The present invention, in its preferred embodiment, utilizes the SIP and RTP protocols, it being understood that in alternative embodiments, the present invention may be implemented on other analogous systems or environments or using analogous protocols.

The present invention has been described herein with regard to preferred embodiments. However, it will be obvious to persons skilled in the art that a number of variations and modifications can be made without departing from the scope of the invention as described herein.

Claims

OR PRIVILEGE IS CLAIMED ARE DEFINED AS FOLLOWS:

1. A method for half-duplex communication between a first SIP Communication Device having access to a computer network and a second SIP Communication Device having access to that computer network, comprising the steps of: a. establishing a first RTP channel between the first SIP Communication Device and a first computer on the network; b. establishing a second RTP channel between the second SIP Communication Device and a second computer on the network; c. establishing a communication channel between the first computer and the second computer; d. limiting transmission access on the communication channel between the first computer and the second computer to a maximum of one computer at a time.

2. The method according to claim 1 wherein the step of establishing the first RTP channel includes the steps of:

a. the first SIP Communication Device transmitting a request to the first computer to initiate an RTP session between the first SIP Communication Device and the first computer; and b. the first computer transmitting a response back to the first SIP Communication Device; c. the first SIP Communication Device moving to the connected state.

3. The method according to claim 2 wherein steps 2(a) and 2(b) utilize Session Initiation Protocols.

4. The method according to claim 1, 2 or 3 wherein the step of establishing the second RTP channel includes the steps of: _ computer to initiate an RTP session between the second SIP Communication Device and the second computer; and b. the second computer transmitting a response back to the second SIP Communication Device; c. the second SIP Communication Device moving to the connected state.

5. The method according to claim 4 wherein steps 4(a) and 4(b) utilize Session Initiation Protocols.

6. The method according to claim 1, 2, 3, 4, or 5wherein the step of limiting transmission access to the communication channel may be taken by any computer with access to the established communication channel.

7. The method according to claim 1, 2, 3, 4, 5 or 6 wherein the step of limiting transmission access to the communication channel may be taken by the first computer.

8. The method according to claim 1 , 2, 3, 4, 5, 6 or 7 wherein the step of limiting transmission access to the communication channel may be taken by the first computer in response to a PUSH TO TALK signal the first computer receives from the first SIP Communication Device.

9. The method according to claim 1, 2, 3, 4, 5, 6, 7 or 8 wherein the PUSH TO TALK signal utilizes at least one RTP protocol.

10. The method according to claim 1, 2, 3, 4, 5, 6, 7, 8 or 9 wherein a. the first computer has associated therewith: i. a first buffer adapted for communication with the first SIP Communication Device for temporarily storing data transmitted from the first SIP Communication Device; ii. a second buffer adapted for communication with the first SIP first SIP Communication Device; iii. a third buffer adapted for communication with the network for temporarily storing data for transmission to the network; iv. a fourth buffer adapted for communication with the network for temporarily storing data transmitted from the network; v. a first deactivatable communication link between the first buffer and the third buffer which communication link defaults to the deactivated mode; vi. a second communication link between the second buffer and the fourth buffer; vii. a fifth buffer adapted for communication with the network for temporarily storing RTCP data for transmission to the network; and viii. a sixth buffer adapted for communication with the network for temporarily storing RTCP data transmitted from the network; and the second computer has associated therewith: i. a first buffer adapted for communication with the second SIP

Communication Device for temporarily storing data transmitted from the second SIP Communication Device; ii. a second buffer adapted for communication with the second SIP

Communication Device for temporarily storing data for transmission to the second SIP Communication Device; iii. a third buffer adapted for communication with the network for temporarily storing data for transmission to the network; iv. a fourth buffer adapted for communication with network for temporarily storing data transmitted from the network; v. a first deactivatable communication link between the first buffer and the third buffer which communication link defaults to the deactivated mode; vi. a second communication link between the second buffer and the fourth buffer; vii. a fifth buffer adapted for communication with network for temporarily storing RTCP data for transmission to the network; storing RTCP data transmitted from the network.

11. A method according to claim 10 wherein the step of limiting transmission access to the communication channel by the first computer includes the steps of: a. activating the first deactivatable communication link between the first buffer and the third buffer associated with the first computer; b. transmitting to the network by way of the fifth buffer associated with the first computer a command by way of an RTCP APP packet; c. receiving on the second computer by way of the sixth buffer the command by way of the RTCP APP packet; and d. locking out the activation of the first deactivatable communication link between the first buffer and the third buffer associated with the second computer.

12. A method according to claim 10 wherein the step of limiting transmission access to the communication channel by the second computer includes the steps of: a. activating the first deactivatable communication link between the first buffer and the third buffer associated with the second computer; b. transmitting to the network by way of the fifth buffer associated with the second computer a command by way of an RTCP APP packet; c. receiving on the first computer by way of the sixth buffer the command by way of the RTCP APP packet; and d. locking out the activation of the first deactivatable communication link between the first buffer and the third buffer associated with the first computer.