Abstract:
Mobile devices use a PBX and application servers associated with the PBX to access voice services. Some mobile devices can support simultaneous data and voice channels, with the PBX and application server respectively. Where a data channel is unavailable, a control and status updating mechanism is needed. One approach is to signal over the voice channel with DTMF tones. DTMF tones should be timed to cause the tones to arrive at the mobile device during a pause period of a ring cycle. However, it also is desirable to avoid unnaturally long silence periods or other pauses during system usage. Aspects relate to enhancing a user experience in these situations and successful transmission/reception of control and status information over a voice channel using DTMF tones.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This application claims priority benefit of U.S. provisional application No. 61/328,146, filed on Apr. 26, 2010. U.S. provisional application No. 61/328,146, is fully incorporated by reference herein. 
    
    
     BACKGROUND 
     1. Field 
     The present application relates to voice telephony, and more particularly to control and status signaling to and from mobile devices over voice channels, such as in third party call control. 
     2Related Art 
     Voice telephony remains a major application of interest for business and personal use. In an example corporate setting, a telephony installation at a site can have a large number of users connected to a Private Branch Exchange (PBX) server, which can interface those users to a smaller number of outside lines (e.g., an E1 or T1 line) (a PBX can be implemented using a server with telephony cards for example). A PBX can interface with other servers and systems, such as one or more application servers that can provide enhanced services to devices connected to the PBX, such as mobile devices. For example, a PBX can interface with an application server over an IP connection, using SIP signaling. Services provided by an application server can include voice mail, single number reachability, call forwarding, park, and conferencing, for example. PBX systems also can communicate with each other over trunk lines, and packet networks, depending on implementation. 
     A PBX can perform services for a voice call, based on direction from an application server. For example, a PBX can initiate an outgoing call to a number specified by an application server using SIP signaling to the PBX, and bridge a mobile device that requested such call. Continuing this example, it would be desirable to provide status information to the requesting mobile device. Such status information would include information such as whether the remote party phone has started to ring or not. Still further, initiating commands, such as call transfer, and sending status information during progress of such commands also desirably is available. In absence of a data channel that can be maintained concurrently with a voice channel, this information can be communicated using DTMF tones. However, a number of considerations desirably should be addressed if using DTMF tones for such purposes, and aspects herein relate to alleviating some of these concerns. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Reference will now be made, by way of example, to the accompanying drawings which show example embodiments of the present application, and in which: 
         FIG. 1  shows, in block diagram form, an example arrangement in which voice calls involving mobile devices can be setup, and which involves an enterprise communications platform; 
         FIG. 2  depicts a first method that can be implemented on a mobile device for use in synchronizing ring tone cycles with DTMF tone signaling; 
         FIG. 3  depicts a second method that can be implemented on a mobile device for use in synchronizing ring tone cycles with DTMF tone signaling; 
         FIG. 4  depicts a server counterpart method for the method of  FIG. 3 ; 
         FIG. 5  depicts a server counterpart method for the method of  FIG. 2 ; 
         FIG. 6  depicts an example composition of a mobile device that can perform aspects in accordance with this disclosure; 
         FIG. 7  depicts another view of an example of a mobile device that can perform aspects in accordance with this disclosure; and 
         FIG. 8  depicts a functional module composition of a mobile device in accordance with this disclosure. 
     
    
    
     DESCRIPTION 
     In many cases, modern mobile devices, such as phones, smartphones, and other network-enabled devices can access both a data network and a voice network (depending on technology, access of each network type may occur concurrently or interleaved). It is becoming increasingly desirable, especially for corporate users, to be able to use their mobile phones in a manner similar to their deskphones, such as having similar functionality, one number reachability, and so on. Such capabilities can be provided to mobile devices through an application server that can send and receive status information and control commands over a data channel with the mobile device. If a data channel is not currently available, DTMF tones can be sent over a voice channel for these purposes. 
     Some mobile devices have trouble detecting DTMF tones during a ring portion of a ring tone cycle that the device is locally generating. So, it is desirable to have the mobile device to receive any such tones during a pause period of such a ring tone. 
     A variety of approaches can be implemented directed such timing, and examples of some preferred approaches are described herein. Another consideration is to avoid having a user of the mobile device hear tones being sent for such control and status purposes. 
       FIG. 1  depicts an example arrangement, which will be used in the following description. In some aspects, this description concerns an approach to reducing a setup time for a call involving a wireless device and carried on a transport that includes elements of the Public Switched Telephony Network (PSTN). The elements of  FIG. 1  are introduced, followed with background to provide context to the particular examples that follow. 
       FIG. 1  depicts an arrangement where a PBX  16  can exist within a corporate network. PBX  16  can be coupled to the Public Switched Telephone Network (PSTN)  40 , such as via a T1 trunk, or through a gateway  13  that interfaces between the PSTN and a packet network technology. Gateway  13  also can convert between packet network signaling, such as H.245 or SIP to PSTN signaling such as ISDN signaling. PBX  16  also can be coupled to the Internet  40 , or to other packet networks, through a firewall  22 , and other network equipment such as routers and switches (not separately depicted). PBX  16  also can have a connection to both the PSTN and one or more packet networks. 
     Traditionally, a PBX (e.g., PBX  16 ) provides telephony services for a closed group of private telephones, for example, within an enterprise or a hotel. However, in many cases the services to the private telephone desirably could also be extended out to a mobile device that communicate via third party networks, such as wide area cellular networks, or through wireless local area networks. In these situations SMP server  18  may be added as an adjunct to the PBX and can take an active role in providing services to mobile devices. 
     SMP server  18  can be connected to PBX  16  through a packet-switched connection that can use the Internet Protocol (IP). 
     PSTN  40  can communicate with Public Land Mobile Network (PLMN)  50 , and by particular example, with a switching control  86  within PLMN  50 . Such communication is for accepting calls from PLMN  50  to be carried on PSTN  40 , and vice versa, as an example. SS7 signalling can be employed between PSTN  40  and PLMN  50 . PSTN  40  is depicted to have telephones  87   a  and  87   b  coupled thereto. Other implementations can include multiple gateways that translate between or among different signaling protocols. 
     PLMN  50  communicates with a mobile device  11 . Communication between PLMN  50  and mobile device  11  can take place using a wide variety of technologies, some of which are capable of supporting only voice traffic, either voice traffic and data traffic, or simultaneous voice and data traffic to/from mobile device  11 . PLMN  50  also can use Internet  40  to receive/send traffic to and from corporate network  20 , as indicated by communication link  92 . Communication between network  20  and PLMN  50  also may be carried via a relay  26 . 
     Mobile device  11  includes one or more radio transceivers and associated processing hardware and software to enable wireless communications with the PLMN  50  and optionally a WLAN. In various embodiments, the PLMN  50  and mobile device  11  may be configured to operate in compliance with any one or more of a number of wireless protocols, including GSM, GPRS, CDMA, EDGE, UMTS, EvDO, HSPA, 3GPP, or a variety of others. It will be appreciated that the mobile device  11  may roam within PLMN  50  as well as into other PLMNs (i.e., that the depicted PLMN  50  represents one or more such wireless access networks that operate according to what a person of ordinary skill would understand as broadband cellular access technologies). 
     For example, in some instances, a dual-mode mobile device  11  and/or the enterprise network  20  can be configured to facilitate roaming between the PLMN  50  and a WLAN, and are thus capable of seamlessly transferring sessions (such as voice calls) from a connection with a cellular interface of a dual-mode device to a WLAN interface of such a dual-mode device, and vice versa. 
     GSM signaling  91  can be implemented using the Fast Associated Control Channel (FACCH). FACCH is a logical channel on a digital traffic channel that can be used to send urgent signaling control messages. The FACCH channel sends messages by replacing speech data with signaling data for short periods of time. In GSM, two special reserved bits are used to inform the receiving device if the data in the current time slot is digitally coded subscriber traffic or alternatively a FACCH message. Switching control  86  manages the conversion between GSM signaling  91  and SS7 signaling  90  (for clarity, a network of base stations operating within PLMN  50  is not separately depicted, and usually, there is a connection between switching control  86 , and one or more base station elements, where device  11  can be connected to such base station elements. GSM signaling can be relayed through the base station to a termination point in switching control  86 .) 
     In some embodiments, PBX  16  may be connected to one or more conventional analog telephones  19 . The PBX  16  is also connected to (or part of) the enterprise network  20  and, through it, to telephone terminal devices, such as digital telephone sets  17 , softphones operating on computers  15 , and so on. Within the enterprise network, each individual may have an associated extension number, sometimes referred to as a PNP (private numbering plan), or direct dial phone number. Calls outgoing from PBX  16  to PSTN  40  or incoming from PSTN  40  to PBX  16  can be circuit-switched calls (typically in the absence of gateway  13 ). Gateway  13  also can be considered part of enterprise network  20 , and a boundary of such enterprise network  20  depicted in  FIG. 1  is primarily for convenience. Within the enterprise, e.g. between the PBX  16  and terminal devices, voice calls are increasingly packet-switched calls, for example Voice-over-IP (VoIP) calls. 
     SMP server  18  can perform some aspects of messaging or session control, like call control and advanced call processing features. SMP server  18  may, in some cases, also perform some media handling. Collectively, SMP server  18  and PBX  16  may be referred to as the enterprise communications platform (server), generally designated  14 . It will be appreciated that the enterprise communications platform  14  and, in particular, SMP server  18 , can be implemented on one or more servers having suitable communications interfaces for connecting to and communicating with the PBX  16 , and other network connections. Although SMP server  18  may be implemented on a stand-alone server, it will be appreciated that it may be implemented into an existing control agent/server as a software component comprising instructions configuring a processor, operating with other software components to implement the functionality attributed to it. As will be described below, SMP server  18  may be implemented as a multi-layer platform. 
     The enterprise communications platform  14  implements the switching to connect session legs and may provide the conversion between, for example, a circuit-switched call and a VoIP call, or to connect legs of other media sessions. In some embodiments, in the context of voice calls the enterprise communications platform  14  provides a number of additional functions including automated attendant, interactive voice response, call forwarding, voice mail, etc. It may also implement certain usage restrictions on enterprise users, such as blocking international calls or 1-900 calls. In many embodiments, Session Initiation Protocol (SIP) may be used to set-up, manage, and terminate media sessions for voice calls. Other protocols may also be employed by the enterprise communications platform  14 , for example, Web Services, Computer Telephony Integration (CTI) protocol, Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE), and various custom Application Programming Interfaces (APIs). 
     One of the functions of enterprise communications platform  14  is to extend the features of enterprise telephony to the mobile devices  11 . For example, the enterprise communications platform  14  cay allow mobile device  11  to perform functions akin to those normally available on a standard office telephone, such as the digital telephone set  17  or analog telephone set  15 . Example features may include direct extension dialing, enterprise voice mail, conferencing, call transfer, call park, etc. 
     The depicted system may include a number of enterprise-associated mobile devices (device  11  is depicted). Device  11  can be a device equipped for cellular communication through the PLMN  50 , or a dual-mode device capable of both cellular and WLAN communications. 
     Enterprise network  20  typically includes a number of networked servers, computers, and other devices. For example, enterprise network  20  may connect one or more desktop or laptop computers  15  (one shown). The connection may be wired or wireless in some embodiments. The enterprise network  20  may also connect to one or more digital telephone sets  17  (one shown). 
     One approach is estimate start of a local ring tone on a mobile device based on a time stamp in a message received at a server from the mobile device, through a data channel. However, message propagation delay is difficult to determine from such a time stamp. For example, if there is significant delay on the voice channel, and that delay is different from a data channel delay, then the timestamp may not be especially helpful. For example, where a WiFi connection is used for the data channel, a delay on the voice channel quite likely would be different. Therefore in preferred aspects herein, one or more DTMF tones sent from the mobile device to the PBX (server) are used by the server to estimate delay on the voice channel, and consequently when to send a given tone to the mobile device. 
     In a first example, mobile device  11  initiates a call (such as by sending a notification or invite over a data channel) received by server  18 . In this situation, mobile device  11  sends verification tone(s) after the cellular portion of the voice channel is established. Where synchronization of ringing is to occur, mobile device  11  also restarts a local ring tone (ringback tone) that is audible on mobile device  11 . Thus, server  18 , when receiving the verification tone (or if tones, then preferably the first tone received) serves as a marker for when mobile device  11  restarted the ring cycle. Using this basis for determining the start of the ring cycle, server  18  then times sending of the DTMF tones indicative of status or control information to arrive at mobile device  11  during ring off periods. 
     The verification tone(s) may need to be sent repeatedly. In one preferred example, when repeatedly sending verification tones, different tones can be used for each, or a cycle of verification tones can be used. In this way, server  18  also is provided with information about when mobile device  11  restarted its ring cycle. An example of such a preferred approach is that a first set of verification tones to be sent are DTMF tones for the numerals 2, 3, 4, 6, and 0. Three seconds after sending theses initial tones, tones for numerals 1, 7, 8, 9, and 0 are sent. If the verification tone continues to need to be repeated, then 3 seconds later, the original tones for numerals 2, 3, 4, and 6 are again transmitted, and so on with the second set of digits sent, if necessary. This example is more appropriate for situations where a ring cycle has a time period roughly modulo  6 . In this way, server  18  can determine based on which digits are received, a modulo value representative of when mobile device  11  restarted its ring cycle. 
     Upon having an estimate as to the timing of a ring cycle occurring on device  11 , server  18  also can further select a time within the pause period to target. In many typical cases, the pause period is 4 seconds. In one preferred case, where the verification sequence is received by server  18  (tones for numerals 2, 3, 4, 6 in the above example), a guard time after an estimated start of the pause cycle and before an estimated end of the pause cycle is around 50 ms. For a second verification tone sequence (1, 7, 8, 9 in the above example), a guard time can be selected as 100 ms. Guard times can be selected based on network and device specific measurements and feedback from users. 
       FIGS. 3 and 4  depict another example where server  18  can initiate the voice channel to mobile device  11 , and considerations that occur in this scenario.  FIG. 3  depicts a device  11  method, which includes that the device  11  can send an indication that a voice channel is to be established ( 216 ), and after a delay (e.g., 1.5 s) a local ring tone is started ( 218 ), Subsequently, device  11  receives a call (request to establish a voice channel) ( 220 ), and once the channel is established, device  11  sends a primary verification tone sequence ( 222 ) and restarts the local ring tone cycle ( 224 ). If an acknowledgement is not received within a timeout period, a secondary verification tone sequence is sent ( 234 ). The primary and secondary verification tone sequences can correspond to the numerical sequences described above. If an acknowledgement is received (either at  226  or at  230 ), then the call can proceed ( 232 ). If not, then the primary verification tone sequence is sent again ( 222 ) after a pre-determined delay. A second concurrent call can be initiated, such as by depicted process ( 233 ), which can be composed of sending ( 216 ), starting local ring tone ( 218 ) and stopping the local ring tone, after sending ( 216 ) and waiting for arrival of a remote ring tone status update. 
       FIG. 4  depicts a server-side method to correspond with the method of  FIG. 3 . Server  18  receives voice call setup information ( 240 ) and establishes a voice channel to device  11 . After the channel is established, server  18  waits to receive DTMF tones on the voice channel ( 244 ), and upon receiving such tones, server  18  compares the tones received with a primary verification code sequence, and if matching, then server  18  can estimate ( 250 ) a start of the ring tone cycle on device  11  based on this received sequence. If tones were received, but do not match the primary sequence, then server  18  compares the tones received to a secondary verification tone sequence ( 248 ), and if matching, server  18  estimates ( 252 ) a timing of the ring cycle on device  11  accordingly. In one example, there is known, pre-defined delay between when device  11  would send the primary and secondary verification tone sequences, such that estimates in ( 252 ) and ( 248 ) can be conducted accordingly. Further DTMF tones sent are timed ( 254 ) based on the estimates made in either  252  or  248 . 
     In other call situations, mobile device  11  may receive a trigger tone indicating whether or not an authentication code (previously provided to mobile device  11 ) should be sent on the voice channel or not. Even if mobile device  11  is not to respond with the full authentication code, it still will respond with an acknowledgement tone(s). Using the example of sending the authentication code, mobile device  11  preferably sends tones for the numerals 0, 1 and the authentication code. Mobile device  11  also generally concurrently with sending these tones also restarts the local ring tone. Server  18  uses these tones, when received, as an indication when mobile device  11  restarted its ring tone, and bases timing decisions for tones to be sent to mobile device  11  from this estimated ring cycle restart time. 
     In these situations, there is some delay on the voice channel, which preferably is determined empirically, and can vary depending on how mobile device  11  is constructed, and the network or networks that it uses. 
     In the situation where there is a trigger tone being sent from server  18  to mobile device  11 , there is no synchronization strategy in place when that trigger tone is to be sent to mobile device  11 . However, it also is preferably that a user of mobile device  1  does not hear the trigger tone. As such, preferably mobile device  11  stops its local ring tone cycle when the cellular leg to the device is connected, so that it can detect the trigger tone, but stays muted so that the user will not actually hear this tone. So similar to the previous example, mobile device  11  can send an invite on the data channel, wait a period of time (e.g., 1.5 seconds), and then begin generating a local ring tone. Upon connection of the cellular leg of the voice channel, mobile device  11  would stop the local ring tone. After getting a trigger tone, and responding, mobile device  11  would restart the local ring. Preferably, mobile device  11  also introduces a delay of another 100 ms to avoid allowing the user to hear a tail portion of the trigger tone. 
     However, if the trigger tone is not received timely, a long period of silence would transpire, while the trigger tone is resent (retried) from server  18 . In one example, the ring cycle is 4 seconds off, 2 seconds on. 
     Situations where retried trigger tones may be heard include where a first trigger tone or the response to that first trigger tone was lost. If the first trigger tone was lost, then mobile device  11  would not have restarted ringing, but if the response was lost, then the device would have restarted ringing, but server  18  would not have received the response and would not be able to confirm a start time of such ringing. 
     Under these circumstances, server  18  will assume that the trigger tone did reach device  11 , and that ringing was restarted. Server  18  will estimate when the ringing started based on when it sent the trigger tone, plus a processing delay. Based on a known pattern of the ring tone, it will schedule further tones with a goal to make the tones arrive at what it estimates to be a pause period. A timeout or numerical limit to retry the trigger tone can be enforced (such as 3 times). 
       FIGS. 2 and 5  depict example methods relating to a device calling into a PBX.  FIG. 2  depicts that mobile device  11  can indicate to server  18  that a voice channel is to be established with device  11 , and can receive data, such as a DID number to call and an authentication token. Device  11  can thereafter initiate establishment of the voice channel to a DID number ( 203 ) (e.g., such as one provided from server  18 ). After a wait period (e.g., 1.5 seconds), begins ( 204 ) a locally audible ring tone cycle at device  11 . Subsequently, a cellular leg of the call is found to be established ( 206 ), and responsively, the local ring tone is ceased ( 208 ). Device  11  waits for one or more tones over the data channel ( 210 ), and when received, restarts the local ring tone ( 212 ) and responds with one or more response DTMF tones ( 214 ). The restarting ( 212 ) and responding ( 214 ) should occur within a functionally short interval, so that server  18  can estimate when device  11  restarted the ring tone cycle based on when it receives the response tones sent in  214  (and on measures and/or estimates of delay on the voice channel). An amount of time between  212  and  214  may vary among implementations, but preferably,  212  and  214  occur within a short period of time, such as 100 ms. 
       FIG. 5  depicts a server  18  counterpart method for the method depicted in  FIG. 2 . In  FIG. 5 , server  18  receives ( 260 ) voice call setup information, such as SIP invite. Subsequently, a voice channel is established to device  11 , such as by device  11  calling a DNIS number available to PBX  16 . Server  18  then causes a trigger tone to be sent ( 264 ) on the voice channel. When a response comprising one or more tones is received ( 266 ), the timing of when the tones are received is used to estimate ( 268 ) the timing of when device  11  restarted its local ring cycle, and timing of further tones to be sent to the device are based on this time estimate. If the response was not received, then server  18  assumes that the trigger tone was received, but that the response to it was lost, and estimates a timing of the ring tone cycle based on an estimate of a round trip delay to and from device  11 , plus some processing delay at device  11 , and times a subsequent trigger tone to be sent ( 264 ) accordingly. A retry limit of trigger tones can be enforced, such as a limit of 3 times. 
     The above description applies to initiating a first voice connection with a given mobile device, e.g., device  11 ), if device  11  wants to make another call, then the authentication/verification procedure can be dispensed, since the existing voice channel has served that end. Thus, for subsequent services, the following aspects can be implemented for synchronizing tones with ring cycles. A subsequent call can be signaled via DTMF or via a data channel, if available. 
     In DTMF case, device  11  sends 2nd outgoing call command digits to server  18  one digit at a time. Once device completes sending the last digit of the 2nd outgoing call feature command, it stops a local ring tone, and waits for arrival of a remote ring tone status update. Server  18  sends this status update when ring synchronizing is required. Device  11  restarts its local ringing cycle when it receives the remote ring status message. Server  18  can estimate when device  11  restarted or started its local ring based on when it sent the remote ring status message. A timestamp for when it sent the message can be recorded, and estimates of network delay and processing time can be added to that time in order to estimate when the local ring at device  11  started. 
     Device  11  may not always detect or receive the remote ring status message. Device  11  however can restart ringing after a delay selected based on an expected delay to receive the remote ringing status message. The device thus should restart ringing based on an estimate of a typical delay to receive such status message. For example, device  11  can wait for X seconds, where X can be selected at 5 seconds, for example. Selecting this time involves a tradeoff between avoiding having the user hear undesirable tones and having a long silence that can make a user wonder what the call status is. 
     Preferably, after receiving remote ring status message device  11  restarts the local ring with a delay to avoid user hearing a tail portion of the trigger tone(s) and to avoid a user hearing an answered status tone when early media is present. For example, when calling into a conference service, early media may connect quickly. One example difference is with a call transfer case, where device  11  will not start ringing unless it receives the remote ring status message. 
     An example of such an approach is found with respect to  FIG. 5 , in which information about a further party to be called can be received ( 272 ), and a call to that party can be made. When the call to the further party begins to ring, a remote ring notification can be sent ( 274 ) to the source of the information received ( 272 ). When further information is to be sent to that device, the local ring cycle on that device (initiated responsive to remote ring notification ( 274 ) is estimated ( 273 ), in order to time delivery of such information, such as by DTMF tones, to arrive during pause periods of the local ring cycle. 
     The term “synchronization” has been used to refer to estimating timing of DTMF tone delivery to coordinate with a ring cycle on a mobile device (e.g., in order to increase a likelihood that a tone sent would arrive during a pause or ring off period of the tone cycle). The usage of synchronization does not imply or require any absolute timing relationship or that there is a guarantee that tones will be delivered during an off period, but rather more generally refers to approaches that heuristically attempt to increase the likelihood of causing tones to arrive at appropriate times at a mobile device, for example. 
       FIG. 6  depicts example components that can be used in implementing a mobile transceiver device  11  according to the above description.  FIG. 2  depicts that a processing module  721  may be composed of a plurality of different processing elements, including one or more ASICs  722 , a programmable processor  724 , one or more co-processors  726 , which each can be fixed function, reconfigurable or programmable, one or more digital signal processors  728 . For example, an ASIC or co-processor  722  may be provided for implementing graphics functionality, encryption and decryption, audio filtering, and other such functions that often involve many repetitive, math-intensive steps. Processing module  721  can comprise memory to be used during processing, such as one or more cache memories  730 . 
     Processing module  721  communicates with mass storage  740 , which can be composed of a Random Access Memory  741  and of non-volatile memory  743 . Non-volatile memory  743  can be implemented with one or more of Flash memory, PROM, EPROM, and so on. Non-volatile memory  743  can be implemented as flash memory, ferromagnetic, phase-change memory, and other non-volatile memory technologies. Non-volatile memory  743  also can store programs, device state, various user information, one or more operating systems, device configuration data, and other data that may need to be accessed persistently. 
     User input interface  710  can comprise a plurality of different sources of user input, such as a camera  702 , a keyboard  704 , a touchscreen  708 , and a microphone, which can provide input to speech recognition functionality  709 . 
     Processing module  721  also can use a variety of network communication protocols, grouped for description purposes here into a communication module  737 , which can include a Bluetooth communication stack  742 , which comprises a L 2 CAP layer  744 , a baseband  746  and a radio  748 . Communications module  737  also can comprise a Wireless Local Area Network ( 747 ) interface, which comprises a link layer  752  with a MAC  754 , and a radio  756 . Communications module  737  also can comprise a cellular broadband data network interface  760 , which in turn comprises a link layer  761 , with MAC  762 . Cellular interface  760  also can comprise a radio for an appropriate frequency spectrum  764 . Communications module  737  also can comprise a USB interface  766 , to provide wired data communication capability. Other wireless and wired communication technologies also can be provided, and this description is exemplary. 
     Referring to  FIG. 7 , there is depicted an example of mobile device  11 . Mobile device  11  comprises a display  812  and a cursor or view positioning device, here depicted as a trackball  814 , which may serve as another input member and is both rotational to provide selection inputs and can also be pressed in a direction generally toward housing to provide another selection input. Trackball  814  permits multi-directional positioning of a selection cursor  818 , such that the selection cursor  818  can be moved in an upward direction, in a downward direction and, if desired and/or permitted, in any diagonal direction. The trackball  814  is in this example situated on a front face (not separately numbered) of a housing  820 , to enable a user to maneuver the trackball  814  while holding mobile device  11  in one hand. In other embodiments, a trackpad or other navigational control device can be implemented as well. 
     The mobile device  11  in  FIG. 8  also comprises a programmable convenience button  815  to activate a selected application such as, for example, a calendar or calculator. Further, mobile device  11  can include an escape or cancel button  816 , a menu or option button  824  and a keyboard  820 . Menu or option button  824  loads a menu or list of options on display  812  when pressed. In this example, the escape or cancel button  816 , menu option button  824 , and keyboard  829  are disposed on the front face of the mobile device housing, while the convenience button  815  is disposed at the side of the housing. This button placement enables a user to operate these buttons while holding mobile device  11  in one hand. The keyboard  829  is, in this example, a standard QWERTY keyboard. 
       FIG. 8  depicts an example functional module organization of mobile device  11 , and which includes a call module  901 . Call module  901  device  11  includes Media Access Controller(s) and PHYsical media layers that can be managed by the MACs ( 922 ) and a transport protocol(s) module  920  interfaced with the MAC/PHYs. A voice channel processing layer module  918  interfaces with transport control ( 20 ), and receives input from a DTMF tone synthesizer and a speech code  910  that receives input from a microphone  912 . Voice channel processing layer  918  also interfaces with a DTMF tone receiver  905  that can detect DTMF tones sent on a voice channel involving transport control  920  and MAC/PHY  922 . A control module  906  also can interface with each of DTMF tone receiver  905  and synthesizer  904  to both receive information represented by received tones and generate information to be sent using DTMF tones. A UI  914  interfaces with control module  906 . Transport module in turn interfaces with one or more MAC/PHY modules  922 . Collectively, voice channel processing module  918 , transport module  920 , and MAC/PHY module(s)  922  provide a stack for transmission and reception of voice information for a call, as well as command and status information relating to such call. 
     In the foregoing, separate boxes or illustrated separation of functional elements of illustrated systems does not necessarily require physical separation of such functions, as communications between such elements can occur by way of messaging, function calls, shared memory space, and so on, without any such physical separation. As such, functions need not be implemented in physically or logically separated platforms, although they are illustrated separately for ease of explanation herein. 
     For example, different embodiments of devices can provide some functions in an operating system installation that are provided at an application layer or in a middle layer in other devices. Different devices can have different designs, such that while some devices implement some functions in fixed function hardware, other devices can implement such functions in a programmable processor with code obtained from a computer readable medium. 
     Further, some aspects may be disclosed with respect to only certain examples. However, such disclosures are not to be implied as requiring that such aspects be used only in embodiments according to such examples. 
     The above description occasionally describes relative timing of events, signals, actions, and the like as occurring “when” another event, signal, action, or the like happens. Such description is not to be construed as requiring a concurrency or any absolute timing, unless otherwise indicated. 
     Certain adaptations and modifications of the described embodiments can be made. Aspects that can be applied to various embodiments may have been described with respect to only a portion of those embodiments, for sake of clarity. However, it is to be understood that these aspects can be provided in or applied to other embodiments as well. Therefore, the above discussed embodiments are considered to be illustrative and not restrictive.