KR20010101394A - Concurrent ip based voice and data services via wireless networks - Google Patents

Abstract

A problem in the field of mobile communications is that mobile users of wireless networks need to use two transport services to provide an improved voice service as faced by the WAP forum. This is inefficient because it burdens the user and network operator with potentially cost aspects. This problem is addressed by using IP over IP technology, by transmitting individual packets over one circuit switching connection using IP-based data communications, by digitizing and packetizing voice data and control data, It is solved by providing control of the associated enhanced voice service.

Description

Concurrent Internet Protocol-based Voice and Data Services over Wireless Communication Networks {CONCURRENT IP BASED VOICE AND DATA SERVICES VIA WIRELESS NETWORKS}

Today's advanced wireless networks, particularly digital cellular networks such as Pan-European Mobile Communications (GSM) and Code Division Multiple Access (CDMA) schemes, provide users with a variety of delivery services, each with different features and usage costs. While such networks differ in many respects, the present invention may be applied with reference to one particular one, such as GSM, since the present invention is applicable to all such aspects.

GSM provides control information and time slots for user calls within eight time slots for each radio channel structure that has changed time division multiple access (TDMA). In this environment, the channel is a variable that forms an end-to-end transmission path, especially at the operating frequency and at that frequency. By providing a normal capacity of 22.8 kbps with the necessary channel coding in each time slot, it is normally 13 kbps for voice services and 12 kbps for fastest data services. The latter 12 kbps data rate is further reduced to 9600 bps normally by using a radio link protocol that provides redundant data error correction.

The so-called "control channel" (in GSM is actually one of eight time slots per channel rather than an individual channel) is used in mobile devices and call control networks. That is, when a user first enters a network coverage area by picking up the phone to answer or answer a call, it registers a user who is actually on the network. Redundant control channel capacity is used for slow data services. Two slow or narrowband data services are known as Short Messaging Serviec (SMS) or Unstructured Supplementary Serviec Data (USSD). The SMS and USSD data service is actually a slow packet data service because there is space in the communication network so that the message can be sent every time the message is delivered. Voice and high-speed data services that use full time slots are known as circuit switching services because they are set up from the user to the end-point and can be other phones for voice calls or other computers for data calls. .

Users have typically used voice services without the fear or special knowledge of providing complex services. Users simply enter the desired telephone number using the keypad, press the appropriate PROCEED key, and the telephone and network prepare for the rest. When a user is unable to answer a call, they often receive an SMS message from the voice response system of the network in response to an outgoing message of the other user. By using the latest SMS, USSD, and equivalent services, information such as traffic information is transmitted to the user subscribing to the service. However, if the user wants to send a message, there is a need for a computer that interfaces with the SMS or USSD access of the phone or generates and sends the message using basic phone functions by pressing keys in a complicated order. If a user wants to use the circuit switching data service, an integrated or attached computer is needed to handle the data application.

Many data applications have been developed and used over GSM and other networks using conventional Internet-based and other communications, but the Internet protocols (TCP / IP and UDP / IP) are becoming a deficient standard. This is far from optimally using this at present.

The Wireless Application Protocol (WAP) Forum is an industry forum for delivering advanced telephone and information services to users of mobile wireless devices such as telephones, pagers, smartphones and personal digital assistants. The WAP Forum has created a set of specifications to meet these objectives, and is constantly working to complete and strengthen these tasks. The basic concept of WAP is to deliver services using Internet-based technology, users talking to the phone, and related services using a micro browser, the information being similar to the Internet's Internet Protocol (IP) and Hibertext Transfer Protocol (HTTP). Delivered by a communication protocol. The WAP protocol is known as a Wireless Datagram Protocol (WDP) such as UDP / IP over the Internet, WTP (Wireless Transaction Protocol) that provides known transport and optimal segmentation and reassembly, and WSP (Wireless Session Protocol) similar to HTTP.

Between a WAP client and a WAP proxy (the proxy is a server computer acting as an intermediary between the client and the network, in particular the WAP proxy can respond to the transmission of data content in a form independent of the data communication protocol of the network). There is an additional Wireless Transport Layer Security (WTLS) for transmitting authentication and security data. The communication protocols are designed to operate over transmission services available in today's communications networks, such as narrowband SMS and USSD data services and high-speed data circuit switching data services of 9600 bps or less. Content is in the form of Wireless Markup Language (WML) and WML scripts based on the Internet's Extensible Markup Language (XML), Hypertext Markup Language (HTML), and Java Script (WML is an auxiliary and excess set of HTML). WML scripts are an auxiliary set and an excess set of JavaScript). Thus, applications and services can be provided in a WML or WML script without having to worry about which transport service is used, such as SMS, USSD, or circuit swooping. WML content is in the form of a bundle of cards (a set of one or more cards each contain specific portions of WML or WML script content or functionality).

Although the following description of the present invention is referred to as WAP by way of example, similar techniques may be employed using conventional TCP / IP or UDP / IP communications over HTTP sessions and HTML, XML or XML script based applications.

Now consider an enhanced voice-based service, such as a voice mail service, as provided by IBM's DirectTalk products. When a voice message arrives at the user's intelligent voice message service, the message is recorded and the caller's identification is made in several ways unrelated to this discussion (caller ID or voice recognition is used to make this identification. There is an option). The message service generates a suitable WML that describes the various caller identification and call control options (e.g., viewing, discarding, storing, proceeding), so that the user if they want it (PULL model) or not (PUSH model) Send to. The delivery is through one of the transport services available to the WAP client and the WAP proxy, which is responsible for delivering the content reliably, efficiently and reliably to the user. The use of SMS or USSD is advantageous because it can be used during a circuit switched voice call and there is no need to set up the call. When the user responds to a message with a call option that needs to set up a voice call, the WAP-based phone has all the capabilities to set up a call using the WML and internal support WAP library. At a particular time, the user can change the required service function (stop, repeat, etc.) using the phone's microbrowser interface and send the command via its configured SMS or USSD service.

To initiate a voice call in progress, the DTMF function (dual-tone multi-frequency signal is the signal generated by pressing the phone's touch key on a "Touchtone" phone) or some other signal transmission level function. There is a need for the WAP capable phone to implement the interface. The availability is highly dependent on the type of network being used or the equipment provided even if the network structure is provided.

If an SMS, USSD or equivalent service is used for signal transmission during the voice call, the user will pay for two services, the voice call connection and the equivalent service used for the SMS or control signal.

U.S. Application No. 5,799,251 discloses a potential problem with wireless telephone systems in which short data messages transmitted on the control channel during the control signal interfere with the control signal transmission and block the control channel potentially affecting voice traffic. A solution to the problem has been proposed to include reserving a wireless channel for transmitting user data messages, which has reserved a channel that acts as a second control channel. The problem with this solution is the use of network resources caused by the user requiring additional channels and the cost associated with the use.

U. S. Patent No. 5,790, 551 discloses a dynamic allocation of packet data traffic channels that can be used to transmit packetized data based on which channels will be freed in a particular time period, the dynamic allocation channel being separated from the data control channel. do. Therefore, since a dedicated channel is not necessary, the available communication channel can be used more efficiently and flexibly.

The present invention relates to the communication of audio information in a wireless communication network, i.e., a wirelessly linked communication network between some communication devices (e.g., mobile telephones, communication centric PDAs or communication computers) and a network access node (generally referred to as a base station of a cellular network). It is about.

1 is a schematic diagram of a data communication network according to the prior art;

2 is a schematic diagram of a data communication network implementing the invention in accordance with the preferred embodiment;

A first aspect of the present invention provides a method for use in communicating audio information between a wireless communication device and a remote communication device via a wireless communication network, wherein the method includes audio information in a digital form of a per-packet packet in one of the communication devices. Providing call control information in a digital form of individual packets, establishing a call connection between the wireless communication device and a remote communication device, and separate audio packets and individual call control packets through the single call connection. There is a step of transmitting.

The transmission of audio information (voice or synthesized voice data) and control information over one logical end-to-end connection is such that an individual connection is maintained through a call for control information, or an individual connection can establish that call multiple times. There is no need. The present invention eliminates the need to use two transmissions through calls to each communication direction between communication devices, thereby reducing the user's calling cost (in particular, eliminating the need for two connections between the mobile device and the network connection node). To increase the network capacity available to other users. The transmission capacity is used for other SMS and USSD message services because it is not necessary to simultaneously use SMS or USSD low bandwidth or narrowband transmission services or separate high speed transmissions for carrying the call control information, Additional capacity can be made available for control with In addition, reducing SMS or USSD type transport service requests to manage ongoing service control reduces the need to dynamically allocate additional low speed data service capacity.

The present invention has other advantages in addition to using less network resources than conventional solutions. First, in the case of using the present invention, the battery power of the phone consumed due to a call is consumed much more slowly, because only one setting is maintained after the call setup. Second, in an embodiment of the present invention for transmitting call control information in a packet via a circuit switched call connection, the latency is shorter and much more predictable than the latency of a USSD or SMS message. In interactive voice applications, the long latency of SMS is not acceptable and the USSD is high speed, but nevertheless has a longer and unpredictable latency than circuit switched connections. Having the same waiting time for the voice data and the control data is beneficial to itself because the voice data does not have to wait for the control information. For example, if the user selects "FORWARD" as the next message, the users want to move to the next message without delay.

The present invention is contrary to the conventional teachings of the prior art, which have dedicated channels or timeslots for voice data and control data, and for each such data type, each communication type using a communication protocol in which each data type is optimized over time. In other words, the prior art focus is on optimizing voice processing and its associated transport protocols, and in other situations voice services within the limits of the available data capacity of each channel structure standard without focusing on how to reduce voice service costs. To improve. This can be understood in terms of data communication capacity that can be used in significant development efforts to keep pace with conflict requirements for high sound quality, extensive error correction, and the like.

According to a preferred embodiment of the present invention, the step of establishing a call connection follows the user of the calling device (e.g., mobile phone) requesting access to the messaging center using the interactive messaging service, in turn, PUSH notification of new messages, regular Follow one of the user or mobile device initialization checks (contents PULL). Establishing the connection sends the first notification from the calling device (eg via SMS or USSD) to the called device (eg the messaging device) and the intent of the calling device to establish an interactive voice directed section. It is advisable to notify the called device and then to the called device and the caller device to establish the connection (possible after sending the approval to the caller). Upon establishing and identifying this connection, the devices inform each other that the new channel will be used for the next transfer of control information. Alternative and simple methods that include the same steps in addition to omitting the initial notification step are equally acceptable.

After establishing the connection, only one call connection is needed for the transmission of voice data and associated progress control information. The voice data packet and the control packet are transmitted using a common transport protocol. This may be a protocol based on the current WAP protocol on IP, for example using WML or WML script in data content format. At the end of the interactive session, the calling device may indicate the end of the session or simply terminate the connection, one of these actions being a condition for resuming shared low bandwidth transmissions for subsequent notification and access. It is defined in advance.

Voice communication requires the delivery of certain content, but people are tolerant of minor obstacles in the language, such as "holes" and clicks, in which a small amount of information is not conveyed. The present invention preferably transmits language content and control information in separate packets via a single circuit switching connection that allows for less complex and lower cost regular delivery. A similar approach on packet networks reduces errors and packet loss, including controlling voice to some buffers while reducing control through regular delivery and contention so that the control information is small enough to fill the gap at buffer management time. You can recover.

The present invention preferably uses " voice over IP " which digitizes the voice information and transmits it over the network of individual packets using an internet protocol (as opposed to the conventional circuit switching protocol of the telephone network). , Circuit switching connections are preferred). However, the present invention is not limited to specially performing "voice over IP", but includes means for carrying a suitably encoded voice over an IP based connection.

In a network architecture such as GSM, which divides a communication channel into a plurality of timeslots, the present invention provides a non-standard data transmission combined with suitable coding of voice information with extra capacity remaining in one timeslot or providing extra capacity. When using speed, it provides the best advantage of providing voice and control information within one time slot capacity.

In a second aspect of the invention, a wireless communication device (such as a mobile telephone or a PDA equipped with a communication facility) includes means for encoding digital information in digital form into individual packets, means for generating individual packets of digital call control information; Means for establishing a call connection through an access node of a communication network, means for transmitting the individual audio packet and an individual control packet to a local access node via the call connection, and an individual packet including audio information and call control information. Means for receiving an individual packet comprising from said network access node via said call connection, and means for decoding said received audio information packet and control information packet.

A third aspect of the present invention provides a messaging center for providing a voice-based communication service in a communication network supporting a wireless communication network, wherein the communication network receives an signal from a mobile telephone in a local cell and transmits the signal to the mobile telephone (or A base station; means for digitally encoding audio information into individual packets, means for generating individual packets of digital call control information, and a call connection between a messaging center and a mobile telephone within said communication network. Means for setting the means, means for transmitting the encoded individual audio packet and the individual control packet to the mobile telephone via the call connection, and the individual packet and call control information with audio information transmitted from the mobile telephone across the network. Means for receiving individual packets with the received audio information And it means for decoding the packets and control information packets.

The messaging center includes means for transmitting a first message having one or more operation selection menus in the message content to a wireless communication device via a wireless communication network, and receiving a message transmitted in response to the first message from the wireless communication device. Means for establishing the call connection while providing a selected action in response. Thus, the messaging center may unify messaging to deliver a plurality of other data content types (email, voice, e-commerce, data lookup operations) in a request made using a selectable menu delivered in WML, WML script or other same language. Providing a service, the response to such an action may be converted to a voice format using sentence-to-speech technology.

A wireless communication network according to the prior art (ie, a communication network including a wireless link) is schematically shown in FIG. Mobile phone 10 is shown in communication with remote messaging center 50 via two other transmission paths 70, 80. The telephone communicates with a communication network 30 via a network access node 20. The messaging center transmits / receives voice data and provides a voice service such as a voice mail to record when it is unable to contact the user's telephone (eg, switched off by other call or out of range of a network access unit). Is one example. The communication between the mobile phone and the messaging center and one of the other communication devices is via an SMS or USSD alert message via a Short Message Service Center (SMSC) or an unstructured supplementary services data center (USSDC) 40. And a first connection established for transmitting the control information in the form of. A second connection over the other transmission path is established in response to the alert, which is used for voice data communication between the mobile telephone and the messaging service or other communication device. The messaging service and communication is via a conventional Interworking Function (IWF) 60 which converts between data transmission protocols and wireless communications on a fixed telephone line. In addition to the alert message being used to trigger a second connection establishment for transmitting voice data, this first connection will be used for subsequent transmission of call control information. The WAP communication environment 90 provides control data access to various transmissions, including SMS, USSD, and the like. Conventional voice access (eg, PSTN or ISDN) is used to access common messaging services.

This conventional device has the problem that even if there is free space in each connection, two separate connections must be maintained through the call. The use of such network resources is typically charged back to the user.

2 illustrates a communication network suitable for implementing one embodiment of the present invention used to send alerts between mobile phone 100 and messaging center 140 through a short messaging center or an unstructured supplementary service data center 130. do. After receiving the initial alert or after the SMS or USSD response message is returned, a separate call connection 180 is established between the mobile phone and the messaging center via the remote access server or gateway 190. It is evident that the messaging center has a suitable data communication environment for delivering a notification menu that controls the options for managing the message, which can be WAP based.

This individual connection is established, and all subsequent session control information and voice data are transmitted over this connection, and the connection used to deliver the alert is no longer needed. [Of course, if the call connection terminates during a call, a new connection is made. Will be established and the communication path may be different than reconnection, for example the mobile phone moves to a new call of the cellular network. However, this should not cause any problem, such that a new connection is made by a new identification (IP address) notification to the mobile phone].

Transmission of voice and control data over a connection includes voice signals (digitization, encoding including compressing and packetizing the audio data) so that the voice and control data can be carried across a common communication protocol. do. The replacement of the initial control channel by the one voice and control connection is notified to communicate with the device that is part of the process of establishing the voice connection and to terminate the initial control connection.

When the main interaction between the mobile phone and the messaging center is no longer needed, the call is suspended and all control passes back to the initial control path 170.

The roles of the various components of a communication network implementing the invention in accordance with the preferred embodiment are as follows.

The remote access server accesses the mobile phone to an IP connection. The remote access server provides a path between the mobile communication network and the mobile telephone, depending on whether the actual communication uses packet-based transmission or circuit switching transmission, and the client's IP address, such as Internet code name maritime services, and the IP of other IP facilities. Addresses can be assigned and additional authentication services can be provided. This results in authenticated access.

The short message service center provides a message switching service to the short message service provided in GSM and many other communication networks, so that some 150 bytes of short message are not available when the mobile phone or fixed device is unavailable, or the current message for receiving the SMS. It can be transferred from a mobile phone to a mobile phone or to a fixed location, such as with a selection memory and progression mechanism used when there is no space capacity.

The USSDC provides a message switching service to a USSD similar to that provided by the SMSC for SMS. However, the reliability, no store, and possible USSD features allow simpler high speed transfers than SMD.

The messaging center includes a service environment with a minimal voice message store, and in more complex embodiments may include the entire unified messaging environment. There is a telephone line or an ISDN line for accessing this messaging center from mobile phones and other suitable calling devices, but in this case packetized encoded voice to carry the voice packet as described initially using IP as the communication protocol. This will be used. The messaging center also selects, listens to, moves to, deletes, saves, etc. to listen to your messages, such as old, new, who, etc., to notify and manage the user's interactions with the provided messages. Means for informing, proceeding, and notifying a number of other basic controls, and complex functions such as converting from voice to text, fax, and the like and vice versa. Although a WAP gateway for protocols handling SMSCs is shown in an integrated configuration of message centers, it can be outside of the messaging centers in the same way.

FIG. 2 also shows that the network infrastructure does not change when using existing interworking functions that convert between network specific protocols and mobile network, such as network specific protocols expected outside of PSTN or ISDN. The modified IWF function 200 further includes the IP voice packetization protocol needed to provide a more general voice service to the mobile, removing the voice encoding and making it a plurality of transmission service options, such as conventional circuit switching or high speed. It will replace IP voice just like a normal wireless packet transfer service.

According to a preferred embodiment of the present invention, the calling device such as the message center executes the initial notification by the caller and the called device which set up a call capable of processing voice data traffic and related progress control information in one call. Providing means (eg, via SMS or USSD). Thus, the user requires only one call, and the network has greater SMS and USSD capacity available for other users and services. According to the user's point of view, the cost of using the service is only the cost of using the voice call during the period of using the improved voice service. For the operator, even if the income reduction responds well to the user still using the service in the communication cell when paying for the transmission service beyond reimbursement for the service cost, the performance improvement provided by the present invention is within this period. While SMS or USSD traffic is reduced to satisfy customers more. This is an advantage for other users as well, avoiding the cost of adding additional channels or dedicated SMS, USSD capacity that would potentially increase costs without the necessary income.

Recent developments in voice and Internet technology have purchased a facility to manage the transmission of voice-based information using Internet protocol-based communications. This involves the transmission of voice information in digital form in individual packets rather than using conventional circuit switching protocols in telephony networks. Recently created packet based services can be used (GPRS, etc., arranged by GMS, CDMA packets or UMTS packets, etc.).

The specific mechanism for delivering voice data over IP (hereafter referred to as "VOIP") is derived from the "VoIP Forum", which is a communication protocol standard ITU- with the efforts of major equipment manufacturers including Cisco, VoclTec, 3com and Netspeak. The use of T H.323 has been facilitated. This has become a standard for transmitting audio and video using IP over the general Internet and within intranets. In addition, the forum may facilitate the use of directory service standards to allow users to locate other users and the use of touch tone signals for automatic call distribution and voice mail.

In addition to IP, VoIP uses real-time protocol (RTP) to help ensure that packets are delivered in a timely manner. Using a general network, it is currently difficult to guarantee the quality of service (QoS). Better services will be made possible by private networks managed by businesses or Internet telephony service providers (ITSPs).

The technique used by at least one equipment manufacturer (Netspeak) to help high-speed packet transmission is based on all possible network gateway computers that select the high-speed path before accessing the general network and establishing a TCP socket connection with the other end. To ping the sound.

Businesses using VoIP locate "VoIP devices" (Cisco AS5300 access servers with VoIP features) at the gateway. The gateway receives the packetized voice transmission from a user in the company and then routes it to another area of the intranet (local area network or wide area network), or uses a T-1 or E-1 interface to exchange them with the public exchange. Transmit over the telephone network (PSTN).

For VoIP, voice information is sampled after suitably filtering 8-bit (or more) samples, typically at 8K samples / second, which is 64 kbps or more. This sound quality level is consistent with the telecommunications industry standard which normally uses 8K samples / S with 8 bits / sample. Subsequent encoding in GSM is the recently proposed 6.5 for coding and decoding for speech, which provides its transmission data rate deceleration to less than 13 kbps for full-rate language or to half-rate voice service with acceptable sound quality. Kbps or less Framing it in an appropriately sized IP packet, perhaps paralleling the need to minimize loss of language and recovery using small packets of 64 or 128 bytes / packet, approximately 16- with packets of 64-128 bytes of data. Allows overhead towards the use of large packet results in 19 kbps or 8-9.5 kbps full rate voice for half rate voice.

Standard full rate GSM voice coding and decoding is so fast that it cannot be squeezed into current 9600 bps data calls. However, by increasing to the basic data rate of GSM, the IP encoding full rate rate can be considered. However, the option of a half-rate voice codec with a data circuit of 9600 bps or more can coexist in the same IP communication circuit by applying low bandwidth data such as WAP enhancement service with sufficient redundant data capacity. This is easily made in a calm period of speech, and its appearance may depend to some extent.

As noted above, protocol H.323 was established for the delivery of IP voice-based services. While GSM or other mobile networks using a point-to-point circuit switched call-based form of connection are more complex than absolutely necessary for IP voice, consider GSM's general packet radio service (GPRS) and further, It is more relevant to where there is one option for implementing packetized voice transmission.

Therefore, the problem is solved in the following manner with respect to the above-described improved voice messaging service.

Mobile phone users may not be able to receive some incoming calls for some reason. This may cause the phone to be out of coverage of the network or to be turned off for one of several reasons. The messaging center issues a PUSH alert to the user in the form of a WML deck that is sent to the user via a WAP proxy 150 using an available failure delivery service, such as USSD. When receiving the USSD content, the phone displays it via the micro browser user interface. The user can determine whether this notification is delayed or should take some action on it. Assuming the latter, the user selects to hear the message. Selecting the LISTEN TO MESSAGE operation, the phone selectively transmits in response to the message service via a WAP proxy, and instructs the phone to remote access server gateway for IP access to the WAP proxy and messaging service. You must set up the data call. When establishing an IP connection through the RAS gateway, a message is sent to the message service to establish a new identification if there is an initially transmitted message, and additionally, listen for the command if no message was initially sent. Send to (LISTEN). Once all the identification and commands have been set up, the messaging service can now start to provide the IP voice.

In some respects, the user may block with a new command or the like through an IP data connection to a messaging center via the WAP proxy. At the end of that period, the client sends one final message to the WAP proxy to delete the current temporary RAS identification, recover the bad connection, and then disconnect the call to the RAS gateway.

In order to facilitate practicing the present invention, a set of modifications will now be described which may be made up of the configurations involved in providing WAP mood services over GSM implementing the present invention.

The mobile phone can use the existing speech processing functions for transmit and receive processing (voice encoding and decoding) to obtain the raw raw data rates provided by this method at the available access bit rates. The mobile phone additionally needs the following.

1. Internet protocol communication capacity (eg, UDP / IP) and PPP based data protocol connection establishment in addition to SMS or USSD support for the WAP environment required for the basic service function. This would ideally use IP even in some services that require large amounts of data in the WAP environment, such as "over air programming" or even in basic services where SMS or USSD is not available.

2. Ability to packetize the encoded voice into IP packets for transmission over the connection. There is a need to receive the packetized voice for the received information and send it to the voice decoding circuit of the telephone.

3. Ability to multiplex the voice and data control packets. This capability becomes a normal IP function, but some buffering of the voice packet is provided such that the synchronously (sampled) sampled voice can be transmitted plesiochronously (at normal speed but not synchronously) through the communication network.

The WAP communication function of the messaging center or each gateway will not need any change since the ability to regain connection on another forwarder has already been established.

The messaging center will not need to change its control behavior other than the support of WAP that is normally required, whether or not to implement the present invention. However, the messaging center will require the same functionality as the phone to encode, decode, buffer and manage packetization of voice traffic over IP.

The WAP proxy is provided with the ability to support identity update / change (this is created in the WAP specification). The path between the messaging center, the RAS gateway and the good but less important WAP proxy requires sufficient capacity to avoid congestion if the voice on the IP protocol used on the mobile link does not have enough capacity of H.323, such as a buffer.

Through the above example, the application, i.e., voice over IP, is identified using the set internet principle of the port number, and thus no explicit service identification bit is needed. But. The present invention carries the signal transmission bits in the IP packet carrying the encoded and packetized voice carry other useful information (for quiet periods, encoding algorithms, etc.) to maximize the performance of its options and available carrier capacity.

While addressing the cost issues associated with multiple dedicated channels, each alternative is an alternative method with that problem.

First, we will use DTMF to transmit signals in the audio band. Simple choices can be managed in this way with certainty, but complex messages or other WML messages can be very inefficient. One potential problem is that in some networks that signal during a call such as DTMF, the USSD is used to recover the network, thus gaining no gain and becoming more complex. As presented previously, these features are network type and deployment dependent.

Secondly, the call can be divided into two parts, one being a conventional voice call and the other being a pure data call for the transfer of control information. At a particular time of either the voice or data call, this can reduce the user cost, but becomes more complicated than the present invention and requires much more of the network infrastructure resources even if the number of channels needed at any time is the same as the present invention. Whether the user should charge for one or two calls is a network service provision choice. While the features for supporting voice or data calls at the same time have appeared, the necessary network features are complex and not all networks will have this capacity.

The present invention has already described the use of initial notification over narrow bandwidth transmission and the use of broadband for IP voice service and outgoing control. Alternative implementations using only the packet-based services are possible, but this is undesirable in terms of transmission costs, battery power consumption and communication networks. The solution using packet based services is the same as the network infrastructure than using circuit switched calls, or the RAS server needs to terminate the IP based communication.

Claims (13)

A method for transferring audio information between a wireless communication device and a telecommunication device via a wireless communication network, At one of the communication devices, providing digital form of audio information in individual packets and digital form call control information in individual packets; Establishing a call connection between the wireless communication device and a telecommunication device; Transmitting said individual audio packet and said individual call control packet via said one call connection via a wireless communication network. The method of claim 1, wherein establishing the call connection comprises: Sending a notification from one of the communication devices to the other of the communication devices via a first connection; Establishing the call connection between the devices; Confirming identification of the calling device; Informing each device that subsequent control information is to be exchanged over the call connection. 3. The wireless communication device of claim 2, wherein use of a call connection other than the one call connection comprises informing each of the devices to close the one call connection or restart in response to the termination of a communication work process. And transmitting audio information between the telecommunication device and a telecommunication device through a wireless communication network. The audio information between a wireless communication device and a telecommunication device according to claim 2, wherein the contents of the notification, audio packet and control packet include a WML script and / or a WML, or other XML-based or HTML-based language. The method for transmitting the over a wireless communication network. 5. Audio between a wireless communication device and a telecommunication device according to any one of claims 2 to 4, wherein the notification is transmitted via a low bandwidth bearer transmission service and the call connection uses a broadband transmission service. A method for transferring information over a wireless network. The method of claim 1, wherein the telecommunications device is a messaging center providing voice based communication service, The method, Sending a notification from the messaging center via a first connection to a wireless communication device that can use a message delivered to the wireless communication device; Establishing a connection for voice-based communication by sending a notification from the wireless communication device to a messaging center intended by the wireless communication device; The wireless communication device and the messaging center establishing the call connection; Checking an IDFMF of the communication device; Informing each of the devices that subsequent control information can be exchanged through the established call connection; Closing or suspending the first connection; and transmitting audio information between the wireless communication device and the telecommunication device over a wireless communication network. The wireless communication device according to any one of claims 1 to 6, wherein the individual audio packet and the control packet are transmitted by using an internet protocol through a circuit switched call connection. How to transfer through the network The method of claim 7, wherein the audio packet and the control packet are transmitted by using a protocol ITU-T H.323 through the Internet protocol. Way. The method of any one of claims 1 to 8, wherein providing digital information of audio information in individual packets includes encoding voice data using a coding technique that achieves a transmission rate of approximately 19 kbps or less. Transmitting audio information between the wireless communication device and the telecommunication device via a wireless communication network. 10. The method of claim 9, wherein the encoding using the coding technique achieves a transmission rate of approximately 9.5 kbps or less. In a wireless communication device, Means for encoding digital information in digital form into individual packets and means for generating individual packets of digital call control information; Means for establishing a call connection through an access node of a communication network; Means for sending the individual audio packet and the individual control packet to the local access node via the call connection; Means for receiving from said network access node via said call connection an individual packet comprising audio information and an individual packet comprising call control information; Means for decoding the received audio information packet and control information packet. A messaging center for providing a voice-based communication service in a communication network that supports wireless communication and includes an access node for receiving a signal from a wireless communication device in a local cell and transmitting a signal to the wireless communication device. Means for encoding digital information in digital form into individual packets and means for generating individual packets of digital call control information; Means for establishing a call connection between the messaging center and a wireless communication device in a communication network; Means for transmitting the encoded individual audio packet and individual control packet to the wireless communication device via the call connection; Means for receiving from the wireless communication device an individual packet comprising audio information and an individual packet comprising call control information transmitted via the communication network; Means for decoding the received audio information packet and control information packet. 13. The apparatus of claim 12, further comprising: means for transmitting, via a wireless communication device, a first message to the wireless communication device, the first message comprising one or more menus for selectable actions in the message content; Establishing the call connection to provide a selected action in response to receiving a message sent from the wireless communication device in response to the first message.