Patent Application
20060268848
1. Technical Field
The present invention pertains to telecommunications, and particularly to Voice over Internet Protocol (VoIP).
2. Related Art and Other Considerations
Voice over Internet Protocol (VoIP) in the mobile world means using a packet switched (PS) service for transport of Internet Protocol (IP) packets (which contain, e.g., Adaptive Mutli-Rate codec (AMR) speech frames) for normal mobile phone calls. In circuit-switched networks, network resources are static from the sender to receiver before the start of the transfer, thus creating a “circuit”. The resources remain is dedicated to the circuit during the entire transfer and the entire message follows the same path. In packet-switched networks, the message is broken into packets, each of which can take a different route to the destination where the packets are recompiled into the original message.
The packet switched (PS) service utilized for VoIP can be, for example, GPRS (General Packet Radio Service), EDGE (Enhanced Data Rates for Global Evolution), or WCDMA (Wideband Code Division Multiple Access). Each of these example services happen to be built upon the Global System for Mobile communications (GSM), a second generation (“2G”) digital radio access technology originally developed for Europe. GSM was enhanced in 2.5G to include technologies such as GPRS. The third generation (3G) comprises mobile telephone technologies covered by the International Telecommunications Union (ITU) IMT-2000 family. The Third Generation Partnership Project (3GPP) is a group of international standards bodies, operators, and vendors working toward standardizing WCDMA-based members of the IMT-2000.
EDGE (or Enhanced Data Rates for Global Evolution) is a 3G technology that delivers broadband-like data speeds to mobile devices. EDGE allows consumers to connect to the Internet and send and receive data, including digital images, web pages and photographs, three times faster than possible with an ordinary GSM/GPRS network. EDGE enables GSM operators to offer higher-speed mobile-data access, serve more mobile-data customers, and free up GSM network capacity to accommodate additional voice traffic.
EDGE provides three times the data capacity of GPRS. Using EDGE, operators can handle three times more subscribers than GPRS; triple their data rate per subscriber, or add extra capacity to their voice communications. EDGE uses the same TDMA (Time Division Multiple Access) frame structure, logic channel and 200 kHz carrier bandwidth as GSM networks, which allows existing cell plans to remain intact.
In EDGE technology, a base transceiver station (BTS) communicates with a mobile station (e.g., a cell phone, mobile terminal or the like, including computers such as laptops with mobile termination). The base transceiver station (BTS) typically has plural transceivers (TRX), with each transceiver having plural timeslots. Some of the transceivers (TRX) which may be capable of “hopping”, e.g., frequency hopping. Frequency hopping is a process in which the data signal is modulated with a narrowband carrier signal that “hops” in a random but predictable sequence from frequency to frequency as a function of time over a wide band of frequencies.
A number of situations can result in packet switched (PS) transfer speeds being below what is required for good VoIP quality. One such situation is a drop or decrease in carrier to interference ratio (C/I) to such a low level that additional timeslots (if added) could not compensate for a high bit error rate. Another situation occurs when there is insufficiently allocated capacity to PS data for a specific cell at a specific moment, resulting in “jitter” and too low transfer speed. A third situation is a cell change to an old transceiver (TRX) which is not EDGE-enabled, resulting in a change down to GPRS. A fourth situation is based on limitations in the data network or IP Multimedia Subsystem (IMS) network. A fifth situation occurs when transmission to the RBS site is made with a statistical (packet based) method, resulting in a certain calculated risk of blocking on the actual transmission.
In all these situations, while the VoIP call can survive, speech quality may not be as good as desired. Currently, the IMS system (downlink) and the VoIP client in the phone (uplink) will just keep sending speech data, despite the degree of speech quality, even if the resulting speech quality for the listening party is poor.
A telecommunications network comprises a base transceiver station node and a packet control unit. The base transceiver station node serves, e.g., for providing radio transmission resources to a cell for radio frequency communications. The packet control unit serves for allocating the radio transmission resources to respective voice over internet protocol (VoIP) calls handled as packet switched connections. In addition, for at least one VoIP call, the packet control unit is arranged for determining whether the at least one VoIP call should be changed from one connection type to another connection type, e.g., from a packet switched connection to a circuit switched connection.
In an illustrated, example, non-limiting embodiment, the packet control unit determines whether the at least one VoIP call should be changed from a packet switched connection to a circuit switched connection by monitoring speech quality. In accordance with the monitoring, the packet control unit is arranged for requesting that the at least one VoIP call be changed from a packet switched connection to a circuit switched connection.
In one mode of operation, for monitoring speech quality of the VoIP packet flow for the VoIP call, the packet control unit monitors, in the telecommunications network, a transfer speed of packets comprising the VoIP call. In an example implementation, the packet control unit comprises a buffer and is arranged for monitoring a transfer speed in the buffer of the packets comprising the at least one VoIP. For example, the packet control unit can monitor the transfer speed by determining when a utilized amount of the buffer exceeds a predetermined threshold. Alternatively, the packet control unit can monitor the transfer speed by determining when a variation of a utilized amount of the buffer exceeds a predetermined threshold (e.g., buffer fullness).
In an example implementation, the buffer which is monitored by the packet control unit can be a logical link control layer (LLC) buffer, and the voice over internet protocol (VoIP) calls can be EDGE VoIP packet flows.
In another mode of operation, for monitoring speech quality of the VoIP packet flow for the VoIP call, the packet control unit monitors lost or damaged frames carrying the VoIP speech. If the number of lost of damaged frames exceeds a predetermined limit, the packet control unit requests that the at least one VoIP call be changed from a packet switched connection to a circuit switched connection.
The packet control unit can be located either entirely or partially at any suitable network node, such as at a base station control (BSC) node, the base station node, and a GPRS Support node (GSN).
Requesting that the at least one VoIP call be changed from a packet switched connection to a circuit switched connection can comprises requesting a mobile station participating in the call to perform a packet-switch to circuit-switch handover and thereby reattach the call as a circuit switch call.
The foregoing and other objects, features, and advantages of the invention will be apparent from the following more particular description of preferred embodiments as illustrated in the accompanying drawings in which reference characters refer to the same parts throughout the various views. The drawings are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention.
In the following description, for purposes of explanation and not limitation, specific details are set forth such as particular architectures, interfaces, techniques, etc. in order to provide a thorough understanding of the present invention. However, it will be apparent to those skilled in the art that the present invention may be practiced in other embodiments that depart from these specific details. That is, those skilled in the art will be able to devise various arrangements which, although not explicitly described or shown herein, embody the principles of the invention and are included within its spirit and scope. In some instances, detailed descriptions of well-known devices, circuits, and methods are omitted so as not to obscure the description of the present invention with unnecessary detail. All statements herein reciting principles, aspects, and embodiments of the invention, as well as specific examples thereof, are intended to encompass both structural and functional equivalents thereof. Additionally, it is intended that such equivalents include both currently known equivalents as well as equivalents developed in the future, i.e., any elements developed that perform the same function, regardless of structure.
Thus, for example, it will be appreciated by those skilled in the art that block diagrams herein can represent conceptual views of illustrative circuitry embodying the principles of the technology. Similarly, it will be appreciated that any flow charts, state transition diagrams, pseudo code, and the like represent various processes which may be substantially represented in computer readable medium and so executed by a computer or processor, whether or not such computer or processor is explicitly shown.
The functions of the various elements including functional blocks labeled as “processors” or “controllers” may be provided through the use of dedicated hardware as well as hardware capable of executing software in association with appropriate software. When provided by a processor, the functions may be provided by a single dedicated processor, by a single shared processor, or by a plurality of individual processors, some of which may be shared or distributed. Moreover, explicit use of the term “processor” or “controller” should not be construed to refer exclusively to hardware capable of executing software, and may include, without limitation, digital signal processor (DSP) hardware, read only memory (ROM) for storing software, random access memory (RAM), and non-volatile storage.
The base transceiver station (BTS) 28 serves one or more cells, such as cell 40. In serving cell 40, base transceiver station (BTS) 28 provides a pool 50 of radio transmission resources. As conceptualized in an example embodiment of
In the illustrated, example, non-limiting implementation of
Optionally in the foregoing example implementation, at least one radio transmission resource of the non-hopping set 521 of radio transmission resources can be utilized for a broadcast control channel (BCCH) (and/or for other standardized or common broadcast channels), while other radio transmission resources of the non-hopping set non-hopping set 521 of radio transmission resources can be utilized for calls comprising voice over internet protocol packet flows. For example, at least one timeslot of the non-hopping set 521 of radio transmission resources can be utilized for a BCCH (such as timeslots 561-1, for example), and other timeslots of the non-hopping set 521 of radio transmission resources (such as timeslots 561-2 through 561-j, for example) can be utilized for the calls comprising voice over internet protocol packet flows.
The packet control unit (PCU) 25 comprises resource assignment logic, which can be implemented (for example) by a resource assignment controller 60. In an example embodiment, resource assignment controller 60 schedules calls, the calls taking the form of voice over internet protocol packet flows in the method and/or manner of
For its assignment and allocation of resources, resource assignment controller 60 may include a resource memory 61 or other mechanism for keeping track of allocation or assignment of resources of the sets 52 of radio transmission resources provided by base transceiver station (BTS) 28. The resource memory 61 may resemble a map or image of the sets 52 of radio transmission resources.
In addition, packet control unit (PCU) 25 is arranged and/or configured for determining, for at least one VoIP call handled by packet control unit (PCU) 25, whether the at least one VoIP call should be changed from a packet switched connection to a circuit switched connection. More specifically, in the example embodiment of
Packet control unit (PCU) 25 comprises a buffer for the at least one call and is arranged for monitoring speech quality of the packets allocated to the at least one VoIP. Accordingly, in the illustrated, non-limiting example embodiment of
The pool 70 of packet buffers can optionally be structured or conceptualized, if desired, as sets 82 of buffers, with each set corresponding to one of the set 52 of radio transmission resources provided by base transceiver station (BTS) 28. Thus,
The buffers 84 of pool 70 of packet buffers can be realized or provided in various ways. Each buffer 84 can be a single memory element or device. Alternatively plural buffers 84 can be provided in a common memory element or device, e.g., semiconductor memory device or array, which is addressed, partitioned, or otherwise utilized to store or retrieve data with respect to the plural buffers 84.
In an example implementation shown in
Connection controller 74 governs the particular connection through which a call is made. As such, connection controller 74 implements a connection type for the call, e.g., either circuit switched or packet switched. It is assumed, for a VoIP call, that (at least initially) a packet switched connection is set up by connection controller 74. After the packet switched connection of the VoIP call is set up, the packets forming the downlink packet flow of the VoIP call are routed through an appropriate one of the buffers 84 (the downlink buffer for the call) and packets forming the uplink packet flow of the VoIP call are routed through an appropriate one of the buffers 84 (the uplink buffer for the call).
In an example implementation of
In the example implementation of
Upon its invocation, as step 2-2 the transfer speed monitoring routine checks whether an acceptable transfer speed exists for the VoIP packet switched call for which it was invoked. As such, buffer monitor 72-2 monitors the (e.g., LLC or RLC) buffer fullness in packet control unit (PCU) 25 specifically for VoIP flows.
As mentioned previously, in one example sub-mode of the first mode of operation, the transfer speed monitor 72-2 of packet control unit (PCU) 25 can monitor the transfer speed of packets comprising the VoIP call by determining when a utilized amount of the buffer for the call exceeds a predetermined threshold. Exceeding the predetermined threshold of the buffer tends to indicate that transfer speed has slowed since, e.g., the buffer is filling faster than it is emptying, thereby reflecting reduced transfer speed on the link(s) on the outgoing side.
Alternatively, in another submode of the first mode of operation, the transfer speed monitor 72-2 of packet control unit (PCU) 25 can monitor the transfer speed by determining when a variation of a utilized amount of the buffer (buffer fillness) exceeds a predetermined (e.g. configured) threshold. In this regard, see
In either the two foregoing submodes or other comparable ways of operation, if it is determined at step 2-2 that the transfer speed for the VoIP call is acceptable, the transfer speed monitoring routine (or this instance thereof) can terminate as indicated by step 2-3. Otherwise, step 2-4 is performed.
Step 2-4 is performed when it is determined at step 2-2 that the transfer speed for the VoIP call is not acceptable, e.g., that the transfer speed is slow and therefore that poor speech quality or other low quality or problem occurs. As step 2-4 the transfer speed monitor 72 prompts packet control unit (PCU) 25 to request that the call be changed from one circuit connection type (e.g., a voice over internet protocol packet flow) to another circuit connection type (e.g., a circuit switched connection). Such request can be implemented, for example, by requesting that the mobile station (MS) 30 change the call from a voice over internet protocol packet flow to a circuit switched connection.
Assuming that, in response to the request of step 2-4, the call is switched to a circuit switch call rather than a VoIP call, eventually as step 2-5 the resource assignment controller 60 assigns another radio transmission resource to the (now circuit switched) call. The assigned radio transmission resource is configured or otherwise managed by connection controller 74 as a circuit switched connection. Assignment or reallocation of a call to a circuit switch call is understood by the person skilled in the art and is described, e.g., by section 6.3.6, among others, of 3GPP TS 23.806 V1.7.0 (2005-11), Technical Specification Group Service and System Aspects; Voice Call Continuity between CS and IMS Study (Release 7), incorporated herein by reference in its entirety.
In an example implementation shown in
In the example implementation of
Upon its invocation, as step 3-2 the frame monitoring routine checks whether the number of detected losses or damaged frames thus far noted for the VoIP packet flow (associated with the buffer which it monitors) exceeds a predetermined limit. If not, the frame monitoring routine (or this instance thereof) can terminate as indicated by step 3-3. Otherwise, step 3-4 is performed.
Step 3-4 is performed when it is determined at step 3-2 that the number of detected losses or damaged frames thus far noted for the VoIP packet flow monitored by frame monitor 72-3 exceeds a predetermined limit. Exceeding the predetermined limit is a measure or indication of poor speech quality or other low quality or problem. As step 3-4 the frame monitor 72-3 prompts packet control unit (PCU) 25 to request that the call be changed from one circuit connection type (e.g., a voice over internet protocol packet flow) to another circuit connection type (e.g., a circuit switched connection). Such request can be implemented, for example, by requesting that the mobile station (MS) 30 change the call from a voice over internet protocol packet flow to a circuit switched connection.
Assuming that, in response to the request of step 3-4, the call is switched to a circuit switch call rather than a VoIP call, eventually as step 3-5 the resource assignment controller 60 assigns another radio transmission resource to the (now circuit switched) call. The assigned radio transmission resource is configured or otherwise managed by connection controller 74 as a circuit switched connection. Requesting that the at least one VoIP call be changed from a packet switched connection to a circuit switched connection can comprises requesting a mobile station participating in the call to perform a packet-switch to circuit-switch handover and thereby reattach the call as a circuit switch call. For example, a message is sent from the packet control unit (PCU) 25 to the mobile station (MS) 30 in the form of a “PS-to-CS HO Command”. The mobile station (MS) 30 will perform a PS-to-CS handover and re-attach the call as a circuit switch call on another resources (e.g., on a hopping resource [e.g., a hopping transceiver] or on a non-hopping resource [e.g., a non-hopping transceiver]).
In the non-limiting illustration of
The packet control unit (PCU) 25, for the foregoing and other embodiments encompassed hereby, can be located either entirely or partially at any suitable network node, such as at a base station control (BSC) node 26 as shown in
In an example implementation, the calls comprising voice over internet protocol (VoIP) packet flows are EDGE (Enhanced Data Rates for Global Evolution) VoIP flows. As utilized herein, “EDGE” includes EDGE Evolution, also known, e.g., as EDGE Phase 2.
Both first radio access network 112 and second radio access network 114 are connected to an external core network(s) 116. The core network(s) 116 include a network subsystem 120 for circuit switched connections, featuring a Mobile Switching Center (MSC) 122 which typically operates in conjunction with registers such as a visitor location register (VLR). The network subsystem 120 is typically connected to (for example) the Public Switched Telephone Network (PSTN) 124 and/or the Integrated Services Digital Network (ISDN).
The core network(s) 116 also include a GPRS/backbone 126 which comprises a serving GPRS service node (SGSN) 128 and a Gateway GPRS support node (GGSN) node 130. The GPRS/backbone 126 is connected to connectionless-oriented external network such as IP Network 132 (e.g., the Internet). Thus, the packet switched connections involve communicating with Serving GPRS Support Node (SGSN) 128 which in turn is connected through a backbone network and Gateway GPRS support node (GGSN) 130 to packet-switched networks 130 (e.g., the Internet, X.25 external networks).
The core network(s) 116 can connect to the first radio access network 12 (e.g., the GERAN) over either an interface known as the A interface, an interface known as the Gb interface, or an open Iu interface, or any combination of these three interfaces. In
The core network 116 also connects to the second radio access network 114 (e.g., the UTRAN radio access network) over an interface know as the Iu interface. The second radio access network 114 includes one or more radio network controllers (RNCs) 26U. For sake of simplicity, the UTRAN 114 of
In the particular non-limiting example described in
The foregoing assumes that packet control unit (PCU) 25 can detect a VoIP flow. The person skilled in the art knows how VoIP flow can be detected, e.g., by examining a set of quality of service (QoS) attributes such as (for example) the QoS Conversational bit set by the mobile station in the setup of the VoIP data flow, or by checking for any other type of VoIP signature configured in or appended to the VoIP data flow. Yet other techniques are disclosed in U.S. Provisional Patent Application 60/684,233 entitled “Authenticated Identification of VoIP Flow in BSS,” filed on May 25, 2005, which is incorporated herein by reference in its entirety.
As explained above, step 2-2 and step 3-2 include sending a message from base station controller (BSC) 26 to mobile station (MS) 30 in the form, e.g., of a “PS-to-CS Handover Command”. Such message commands mobile station (MS) 30 to make a packet switched (PS)-to-circuit switched (CS) handover, moving away from the VoIP domain and over/into the traditional CS domain.
Especially in the early phases of VoIP introduction, a number of issues are anticipated since so many components are new. Using the “safe-guard” techniques provided herein, at any issue with the packet switched delivery, no matter the cause, the packet control unit (PCU) 25 will detect the problem condition. Such detection is due, at least in part, in that the number of LLC frames waiting to be sent from the packet control unit (PCU) 25 will be filling up if there is no delivery to the mobile station (MS) 30. Upon such detection of a predetermined amount of waiting frames, the mobile station (MS) 30 will be commanded to handover to the circuit switched domain, where mobile station (MS) 30 is likely to be able to continue the call.
The foregoing technique may facilitate an earlier than otherwise anticipated introduction of voice over internet protocol (VoIP) services.
Buffer (LLC or RLC) level and variation detection as performed in and by packet control unit (PCU) 25 thus enable a commanding of the mobile station (MS) 30 away from the VoIP domain and over/into to the traditional circuit switched domain at any issue that may arise with packet switched delivery. As discussed above, another criteria for discerning poor speech quality (and responsively triggering a packet switch to circuit switch handover for the VoIP flow suffering the poor speech quality) involves monitoring for lost or damaged frames in the VoIP flow.
Although various embodiments have been shown and described in detail, the claims are not limited to any particular embodiment or example. None of the above description should be read as implying that any particular element, step, range, or function is essential such that it must be included in the claims scope. The scope of patented subject matter is defined only by the claims. The extent of legal protection is defined by the words recited in the allowed claims and their equivalents. It is to be understood that the invention is not to be limited to the disclosed embodiment, but on the contrary, is intended to cover various modifications and equivalent arrangements.
This application claims the benefit and priority of U.S. Provisional Patent Application 60/684,214, filed May 25, 2005, the entire contents of which is incorporated by reference in its entirety. This application is related to simultaneously-filed U.S. patent application Ser. No. ______ (attorney docket 2380-921), entitled “CONNECTION TYPE HANDOVER OF VOICE OVER INTERNET PROTOCOL CALL BASED ON RESOURCE TYPE”, which is also incorporated by reference in its entirety This application is related to U.S. patent application Ser. No. ______ (attorney docket 2380-931), filed Nov. 29, 2005, entitled “SCHDULING RADIO RESOURCES FOR SYMMETRIC SERVICE DATA CONNECTIONS”, which is also incorporated by reference in its entirety. This application is also related to the following related US Provisional patent applications, all of which are also incorporated by reference in their entirety: U.S. Provisional Patent Application 60/684,216 entitled “GSM VoIP PS-to-CS Handover at Allocation to Frequency-Hopping Edge TRX,” filed on May 25, 2005; U.S. Provisional Patent Application 60/684,215 entitled “Local Switching AGC,” filed on May 25, 2005; U.S. Provisional Patent Application 60/684,232 entitled “Method to Improve VoIP Media Flow Quality by Adapting Speech Encoder and LQC Based on EDGE MCS; U.S. Provisional Patent Application 60/684,188, filed May 25, 2005; U.S. Provisional Patent Application 60/684,233 entitled “Authenticated Identification of VoIP Flow in BSS,” filed on May 25, 2005.
| Number | Date | Country | |
|---|---|---|---|
| 60684214 | May 2005 | US |
