Source: https://patents.google.com/patent/WO2012063417A1/en
Timestamp: 2018-07-19 00:05:21
Document Index: 532508270

Matched Legal Cases: ['art 610', 'art 708', 'Application No. 2010', 'Application No. 2010', 'Application No. 2011', 'application No. 2011']

WO2012063417A1 - Terminal and coding mode selection method - Google Patents
WO2012063417A1
WO2012063417A1 PCT/JP2011/005986 JP2011005986W WO2012063417A1 WO 2012063417 A1 WO2012063417 A1 WO 2012063417A1 JP 2011005986 W JP2011005986 W JP 2011005986W WO 2012063417 A1 WO2012063417 A1 WO 2012063417A1
PCT/JP2011/005986
江原　宏幸
Provided is a terminal capable of determining the status of a communications path between terminals and selecting an appropriate coding mode without applying a load to either a network or the terminals, by using information held by the terminal as negotiation information. A UE (100) is a terminal that sends to a partner terminal the coding mode selected from a plurality of candidate coding modes notified by the partner terminal, when communications commence between the UE (100) and the partner terminal that is the communications partner. The UE (100) comprises: a wireless receiver (600) that receives a first information relating to the characteristics of an access network to which the partner terminal is currently connected; an information comparison unit (606) that compares the first information and a second information relating to the characteristics of the access network that the UE is currently connected to, and decides the communications format between the UE and the partner terminal; and a mode determination unit (608) that selects the coding mode to be used for communications with the partner terminal, from among the plurality of coding mode candidates and based on the decided communications format.
Terminal and codec mode selection method
The present invention relates to a terminal and codec mode selection method of performing a suitable codec mode selection using the characteristics of the terminal of the destination radio access network (Radio Access Network).
In 3GPP (Third Generation Partnership Project), while IMS VoIP using (IP Multimedia Subsystem) (Voice over IP) service is performed there.
Figure 1 shows an example of a network configuration of VoIP services using 3GPP for IMS. Network shown in Figure 1, an IMS network 126, the IP core network 124 of the operator, the base station: consisting (eNB e Node B) and the eNB radio access network consists of subordinate (Radio Access Network) 120 and 122 Metropolitan . In Figure 1, the terminal (UE: User Equipment) 100,102 are respectively wirelessly connected to the radio access network 120, 122 in ENB104,106, connected to the IP core network 124 via the ENB104,106.
IMS network, the signaling message information management and call control for call control: is a network for performing mutual connection between the routing and 3GPP legacy network or non-3GPP network in the (SIP Session Initiation Protocol).
In IMS network 126 shown in FIG. 1, P-CSCF (Proxy Call Session Control Function) 108,116 is a CSCF in contact with UE100,102. UE, when transmitting the IMS signaling message (SIP REGISTER message, SIP INVITE messages, etc.), searching for P-CSCF that I can connect, and transmits the IMS signaling message to the P-CSCF.
S-CSCF (Serving CSCF) 110,114 is a CSCF for managing management and session contact information of the UE. S-CSCF110,114 is when to manage the contact information of the UE, and download the necessary information from HSS (Home Subscriber Server) 118.
I-CSCF (Interrogating CSCF) 112 holds information CSCF between administrative domains (unit of nets each operator is managed). Etc. When the P-CSCF and S-CSCF has no next node information should be forwarded IMS signaling message, IMS signaling message is transferred via the I-CSCF 112. Also, I-CSCF112 by querying the information in the HSS 118, also confirm the information message transfer destination CSCF. For example, a case will be described in which the SIP INVITE message is sent.
In this case, SIP INVITE message, from the first calling UE, sent via the IP core network to the P-CSCF of the domain (calling domain) to which the UE is present, the calling S-CSCF from P-CSCF It is transferred to. SIP INVITE message, after suitable processing in the calling S-CSCF, is transferred to the S-CFCS the called domain. At this time, there is a case where SIP INVITE message via the I-CSCF 112. Callee S-CSCF is a SIP INVITE message received is forwarded to the called UE via P-CSCF.
IP core network 124 of the operator shown in FIG. 1, the routing of the communication data, QoS (Quality of Service) control, and performs the management of location information of the terminal.
Figure 2 is a flowchart showing an example of a procedure to VoIP calls using the 3GPP IMS is performed. Figure 2 shows a flow example of a case where call the UE102 from UE 100. As shown in FIG. 2, SIP INVITE message is sent from the UE100 through the IMS network to the UE 102 (step (hereinafter, referred to as "ST") 11), SIP 183 Session Progress message is sent from the UE 102 via the IMS network to UE100 that (ST12). Thus, SIP INVITE message and the SIP 183 Session Progress message is exchanged between UE, negotiation is made as to communications.
The SIP INVITE message to the appended SDP (Session Description Protocol) offer mechanism used in VoIP communication is described. The mechanism described, for example, method or codec mode of the codec, and is a candidate concerning protocol. System or codec mode of the codec includes a bit rate, processing delay, the number of channels or the like, the element being employed in the codec. The protocol includes a type of the RTP (Real-time Transport Protocol) payload format. UE102 receives the SIP INVITE message in ST11, select one mechanism among the plurality of candidates described in the SDP offer, described in the SDP answer. UE102 sends the UE100 by adding SDP answer in SIP 183 Session Progress message in ST12.
Selected mechanisms UE102 is analyzed by the IMS network, an instruction for allocating resources in accordance with the analysis result to the call session is issued to the IP core network. According to an instruction from the IMS network, the resource allocation process in the IP core network and the radio access network is performed (ST13). When the resource allocation process is completed, a call is made UE 102 in the user (ST14), user 200OK message is sent to the UE100 and answers (ST15), the call is started between the UE100 and the UE 102 (ST16).
It shows an example of an SDP offer (SDP offer) and SDP answer (SDP answer) in FIG. In Figure 3, the SDP offer, AMR-WB (Adaptive Multi Rate - Wide Band) bandwidth-efficiency mode, AMR-WB octet-align mode, AMR bandwidth-Efficiency mode and the fourth mode AMR octet-the align mode UE100 It is offered in. Moreover, AMR bandwidth-Efficiency mode has been selected in the UE 102.
As for the bandwidth-Efficiency mode and octet-the align mode, is described in Non-Patent Document 1, AMR (or AMR-NB: Narrow Band) is a RTP payload format used and AMR-WB. In the non-patent document 2, the codec bit rate at this time is defined as the AMR 4.75,5.90,7.40, and 12.2 kbps.
Resource allocation processing becomes negotiation fails if resources required in the (ST13 shown in FIG. 2) is not allocated, the again from again transmitting the SIP INVITE message (ST11). Therefore, as the resource allocation is reliably performed, it is recommended to use a range of lower a limited bitrate. For example, Non-Patent Document 3, AMR-NB: 4.75,5.15,5.90,6.70,7.40,7.95,10.2, and 12.2kbps are supported, in Non-Patent Document 4, AMR-WB: 6.60,8.85,12.65,14.25 , 15.85,18.25,19.85,23.05, and 23.85kbps are supported. Non-Patent Document 2, AMR-NB: 4.75,5.90,7.40, and 12.2kbps, AMR-WB: 6.60,8.85, and use of 12.65kbps is recommended.
RFC 4867 IETF, " RTP Payload Format And File Storage Format For The Adaptive Multi-Rate (AMR) And Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs " 3GPP TS 26.114 V9.3.0, " IP Multimedia Subsystem (IMS); Telephony Multimedia; Media handling and interaction "3GPP TS 26.101 V9.0.0," Mandatory speech codec speech Processing functions; Adaptive Multi-Rate (AMR) speech codec frame structure "3GPP TS 26.201 V9.0.0," Speech codec speech Processing functions; Adaptive Rate--Multi Wideband (AMR-WB) Speech Codec Frame Structure "
Codec mode selection by the above-mentioned call start of negotiation, exchange of data is performed independently of the status of the network that is actually performed. Therefore, call start time, or as a codec mode used throughout the call period, there is a problem that appropriate codec mode may not be selected. Currently EVS being discussed by 3GPP SA4 (Enhanced Voice Services), compared with the AMR and AMR-WB, a higher bit rate support, support for multiple processing delay, and supports such as the 2-channel (stereo) are considered there. Therefore, in particular EVS, selection of a suitable codec mode is required.
With respect to the object, UE is the bandwidth of the current Air Interface notifies the server, a method for determining the bit rate to the server based on the bandwidth of the Air Interface has been proposed (for example, "JP-T 2006- see JP "500808). Further, UE will monitor the state of the wireless network, a method of adjusting a suitable de-jitter buffer (de-Jitter Buffer) length according to state of the wireless network has been proposed (for example, "US Patent Application Publication No. 2006 / see Pat "0077994).
However, both methods, minutes for the UE to monitor the radio state (state of bandwidth or radio section of the current Air Interface), is burden to the UE. Moreover, in the above method, (quality of the communication path between UE) end-to-end (End-to-End) communication situation in is not considered.
Further, Non-Patent Document 2, in the negotiation, reliably communicate codec mode regardless communication status (i.e., the possibility of negotiation failure occurs low codec mode) methods of selecting is described. Then, Non-Patent Document 2, further within the scope of the codec bit rate selected by negotiation, describes a method to initiate a call using a low codec bit rate. In this method, after the call initiation, in accordance with the communication status of the end-to-end network (quality of a communication path between UE), adjusting the codec bit-rate within a codec bit rate selected. However, in this method, on can not be set the appropriate codec bit rate (actual codec bit rate suitable for the communication status) from the time the call start, the upper limit of the codec bit rate can be used throughout the call period also is limited.
On the other hand, the network node, and feeds back the communication status of the network (the quality of the communication path between the UE) to the UE, UE is considered a method of selecting a codec mode in accordance with the communication status of the network. However, in this method, the amount of network traffic increases, it takes a burden to the network.
An object of the present invention, while suppressing the processing load on the network and the terminal (UE), a communication terminal and codec mode selection method can be selected from the initial stage of a communication codec mode according to the quality of the communication path between the terminal it is to provide.
Terminal according to an embodiment of the present invention, at the start of communication between the correspondent terminal as a communication partner of the terminal with the terminal, the codec mode selected from the plurality of codec modes candidates notified from the counterpart terminal a terminal to be sent to the other party terminal, and receiving means for the counterpart terminal receives the first information about the characteristics of the access network that are currently connected, the first information and the own apparatus is currently connected comparing the second information about the characteristics of the access network, determining means for determining a communication mode between the own device the counterpart terminal, on the basis of the above communication mode determined by said determining means, said plurality of from the codec mode candidate, it employs a configuration having a selecting means for selecting a codec mode to be used for communication with the counterpart terminal.
Codec mode selection method according to an embodiment of the present invention, at the start of communication between the second terminal is the first terminal a first terminal of a communication partner, the first terminal, the second access in a communication system for transmitting codec mode selected from among a plurality of codec modes candidates notified from the terminal to the second terminal, said second terminal, said second terminal is currently connected and notifying the first information about the characteristics of the network to the first terminal, the first terminal, receiving said first information, said first terminal, said first information When the steps of the first terminal is compared with the second information about the characteristics of the access network currently connected, it determines the communication mode between the first terminal and the second terminal, the through the first terminal, it is determined Based on the embodiment, from the plurality of codec mode candidates, and so includes the steps of: selecting a codec mode used in communication with the second terminal.
According to the present invention, by using as the negotiation information the information terminal is held, without placing a processing load on both the network and the terminal, to determine the status of the communication path between terminals, the appropriate codec mode It can be selected.
It illustrates an example of the IMS network, the IP core network and the radio access network configuration Diagram showing an example of a basic operation of the call session establishment It illustrates an example of an SDP offer and SDP answer It illustrates an example of a configuration of a network according to the embodiments of the present invention It illustrates an example of a configuration of a network according to the embodiments of the present invention Diagram for explaining the RTP packet according to the respective embodiments of the present invention Block diagram showing a configuration of a terminal according to a first embodiment of the present invention (UE) It illustrates an example of an SDP offer according to the first embodiment of the present invention It shows the correspondence between the codec mode sets and numbers according to the first embodiment of the present invention Flow diagram showing the flow of processing in the information comparing unit and the mode determination unit according to the first embodiment of the present invention It illustrates an example of an SDP answer according to the first embodiment of the present invention Block diagram showing the configuration of a network node according to the second embodiment of the present invention Block diagram showing the configuration of a terminal according to Embodiment 3 of the present invention (UE) It illustrates an example of an SDP offer according to a third embodiment of the present invention It shows the correspondence between the codec mode sets and numbers according to the third embodiment of the present invention Flow diagram showing the flow of processing in the information comparing unit and the mode determination unit according to the third embodiment of the present invention It illustrates an example of an SDP answer according to a third embodiment of the present invention It illustrates an example of a format of an IP packet according to the third embodiment of the present invention It shows a number correspondence and RTP payload format of the codec mode according to the fourth embodiment of the present invention (first embodiment) It shows a number correspondence and RTP payload format of the codec mode according to the fourth embodiment of the present invention (second embodiment) It shows a number correspondence and RTP payload format of the codec mode according to the fourth embodiment of the present invention (third embodiment) It shows a number correspondence and RTP payload format of the codec mode according to the fifth embodiment of the present invention (first embodiment) It shows a number correspondence and RTP payload format of the codec mode according to the fifth embodiment of the present invention (second embodiment) It illustrates an example of an operator policy and SDP offer according to a sixth embodiment of the present invention It illustrates an example of an operator policy and SDP answer according to a sixth embodiment of the present invention It illustrates an example of a network configuration according to the seventh embodiment of the present invention Block diagram illustrating a configuration of IP-PBX according to the seventh embodiment of the present invention It shows a first example of a RTP payload format according to the eighth embodiment of the present invention It shows a second example of the RTP payload format according to the eighth embodiment of the present invention It shows a third example of the RTP payload format according to the eighth embodiment of the present invention
Figure 4 is a network configuration according to the embodiments of the present invention, is a diagram showing an example of the positional relationship of the terminal. Incidentally, in FIG. 4, the description thereof are denoted by the same reference numerals with the same configuration as FIG.
HeNB (Home eNode B) 400, 402 are small base stations (here, the femto base station. For example, see "3GPP TS 23.830 v9.0.0," Architecture aspects of Home NodeB and Home eNodeB "". Hereinafter the same) , and the limited CSG ID (Closed Subscriber Group Identifier. for example, see "3GPP TS 25.401 v9.0.0," UTRAN overall description "") indicating that the user can be connected only UE held in each HeNB the Hold. HeNB400 constitute a radio access network 406, HeNB402 constitute a radio access network 408.
In the network configuration shown in FIG. 4, HeNB400,402 is connected to the same Local GW (Gateway) 404. Thus, between UE that is connected to HeNB400 and HeNB402 (UE100 and UE 102) can send and receive without via the Local GW404 data going through the IP core network 124 of the operator.
The radio access network 406, 408 shown in FIG. 4, only a limited user is connected, the radio resource is larger allocatable per 1UE in each radio access network. Such a configuration is applied to, for example, such as when building a dedicated telephone network within the enterprise using a HeNB. In FIG. 4, an example in which Local GW404 is independent of HeNB400,402 are mentioned, but not limited to. Other configurations, for example, Local GW404 is installed to each HeNB400,402, Local GW installed in each HeNB may be configured to adopt connected directly or via another node.
5, a network configuration according to the embodiments of the present invention, showing another example of the positional relationship of the terminal. Incidentally, in FIG. 5, the description thereof are denoted by the same reference numerals with the same configuration as FIG.
HeNB500,502 is a small base station (femto base station), HeNB500 constitute the radio access network 506, HeNB502 constitute a radio access network 508.
In the network configuration shown in FIG. 5, the HeNB500 and HeNB502 are connected respectively separately to the IP core network 124 of the operator. Therefore, HeNB500 and UE each other are connected (UE 100 and UE 102) in HeNB502 transmit and receive data via the IP core network 124 of the operator. However, the radio access network 506, 508, similar to FIG. 4, only a limited user is connected, the radio resource is larger allocatable per 1UE in each radio access network. Such a configuration is applied, for example, when such HeNB in ​​a private home or the like is installed.
In the following description, the communication form (communication mode, each via a connected HeNB and Local GW) between the UE100 and UE102 shown in FIG. 4, referred to as "HeNB + local network communication". Also, the communication mode (communication mode, each via a connected HeNB and IP core network) between the UE100 and UE102 shown in FIG. 5, referred to as "HeNB + IP core network communications."
Here, with reference to FIG. 6, illustrating the RTP-related terms used in the embodiments of the present invention. RTP packet, as shown in FIG. 6, consists of an RTP header and RTP payload.
RTP header are as described in RFC (Request for Comments) 3550 of IETF (Internet Engineering Task Force), which is common irrespective of the type of the RTP payload (codec type, etc.). The format of the RTP payload varies depending on the type of RTP payload. As shown in FIG. 6, the RTP payload is composed of a header portion and a data portion, depending on the type of the RTP payload may not exist header. The header portion of the RTP payload includes information for specifying the number of bits of the encoded data, such as voice and video. The RTP payload data portion, the other encoded data, such as audio and video, the RTP payload length includes padding or the like to be added if the result is not a multiple of 1 octet (8 bits). Further, as shown in FIG. 6, when the RTP packet is transmitted, it is added further lower layer header (UDP (User Datagram Protocol) header, IP header, and IP following transmission path-specific header).
Figure 7 is a block diagram showing the configuration of the terminal (UE100,102) according to the present embodiment. In order to avoid complicated explanation, FIG 7, the present invention and main components involved in the negotiation on the communication between terminals closely related (e.g., components involved in ST11 ~ ST15 shown in FIG. 2 ) shows the.
In the terminal (UE100,102) shown in FIG. 7, the radio reception section 600, if the own apparatus is calling, the signaling message sent from a communication partner destination terminal (the called UE) (SIP 183 Session Progress receiving a message). Then, radio receiving section 600 outputs the signaling messages received mode determination unit 608. This signaling message, for example, described in the SDP answer, codec mode set (bit rate, processing delay, the combination of the number of channels, etc.). Meanwhile, radio receiving section 600, own device when the called party receives a signaling message (SIP INVITE message) sent from the communication partner UE (calling side). Then, radio receiving section 600 outputs the received signaling message to the information comparison unit 606 and the mode determining unit 608. The signaling message, for example, described in the SDP offer, the codec mode set candidate (bit rate, processing delay, the combination of the candidates of the number of channels, etc.), and the radio access network the communication partner of the UE is currently connected It contains information about the characteristics.
Radio transmitting section 602 transmits signaling messages from the signaling generation part 610 (SIP INVITE message or SIP 183 Session Progress message) to the communication partner of the UE.
Destination determination unit 604 determines the characteristics of the radio access network apparatus itself is currently connected, to hold the information about the characteristics of the radio access network. Destination determination unit 604, information about the characteristics of the radio access network, and outputs the information comparing unit 606 and the signaling generator 610.
Information comparing unit 606, if the own apparatus is called side, the own device and information about the characteristics of the radio access network is currently connected, the radio access network the communication partner of UE (calling party) is currently connected comparing the information about the characteristics. The former is the information inputted from the connection destination determination section 604, the latter is inputted from radio receiving section 600, the information contained in the signaling message.
Specifically, the information comparing unit 606 compares the information about the characteristics of both the radio access network, determines the communication mode between the own device and the destination terminal. For example, the information comparing unit 606, the own device and the mating terminal, HeNB + local network communication, HeNB + IP core network communications, and of the different communication modes, to determine which one. Here, HeNB + local network communication is the same HeNB, multiple HeNB or HeNB and VPN (Virtual Private Network) connected communication form in the communication network as configured in the backbone. Further, the HeNB + IP core network communication, a connected communication form HeNB different over IP core network 124 (femto base station). Then, the information comparing unit 606 outputs the determination to the communication form the mode determination unit 608.
Mode determination unit 608, if the own apparatus is calling, the codec mode set that is described in the SDP answer contained in the signaling message received as input from radio receiving section 600, the communication between the own apparatus and the communication partner to determine the codec mode set to be used. Mode determination unit 608, if the own apparatus is called side, the communication from the codec mode sets candidates described in the SDP offer contained in the signaling message received as input from radio receiving section 600, the own device and the destination terminal in selecting a codec mode set that you want to use. This selection is performed based on the form of communication is input from the information comparison unit 606.
Signaling generating unit 610, the own device when the calling party is input from the connection destination determination unit 604 is to generate a signaling message including information about the characteristics of the connection destination access network of its own. Thus, information about the characteristics of the destination access network of the calling UE is notified to the called UE. Signaling generating unit 610, the own device when the called party, generates a signaling message including a codec mode set determined by the mode determining unit 608. Signaling generating unit 610 outputs the generated signaling message to the radio transmitting section 602.
Next, with reference to FIG. 7, the processing in the terminal (UE 100 and UE 102) according to the present embodiment will be described in detail.
In the following description, in FIG. 4 or FIG. 5, current, UE 100 is connected to HeNB400 or HeNB500, UE 102 is connected to HeNB402 or 502, and make a call with respect to UE 102 from the UE 100. The call start time, UE 100 (the calling UE) and UE 102 (the called UE) contains information about what is located where the communication partner with each other (e.g., a cell showing a cell (HeNB) which the UE is connected information, and it does not store the location information, etc.) indicating the position of the UE.
Destination determination unit 604 of the UE100 determines characteristics of the radio access network 406 (or 506) which UE100 is connected. For example, the destination determination unit 604, based on the CSG ID of the UE 100 is currently connected HeNB 400 (or 500), the status of the radio resources allocated to UE 100 (the occupancy of the assignable radio resource or radio resource) to decide. For example, the destination determining unit 604, CSG ID may represent the femto base station, in the femto base station is determined to be large allocable resources to UE 100.
Also as another example, the connection destination determination section 604, which UE100 is the radio access network that is currently connected, the mobile phone dedicated radio access network other than the radio access network such as 3GPP, for example, formed of a wireless LAN or the like it is determined that it is. This determination is made based on such a radio characteristic of a radio access network (radio frequency band or signaling, etc.). In this case, the transfer format of the communication data (the number of frames sent at a time, RTP payload format, etc.), it is determined such that may not be format is limited for mobile phones dedicated radio access network of the 3GPP or the like.
Signaling generating unit 610 of the UE100, the information to be notified to the communication partner (UE 102) is described, the signaling messages for call control (e.g., SIP INVITE message) to generate. For example, when the characteristics of the radio access network at the destination determination unit 604 is determined, the signaling generator 610, when the radio access network currently connected to hold the CSG ID, the CSG ID, included in the SIP INVITE message described in the SDP offer. Or, signaling generating unit 610 may be written to the SIP header (e.g., P-Access-Network-Info header field) (e.g., "3GPP TS 24.229 v10.0.0," Session Initiation Protocol (SIP) and Session Description Protocol see (SDP) "").
Further, as the information about the characteristics of the radio access network, instead of the CSG ID, the base station ID may be used (e.g., "3GPP TS 25.401 v9.0.0," UTRAN overall description "" See). Each UE, based on the CSG ID or base station ID (information about the characteristics of the radio access network), characteristic of the radio access network (e.g., whether a large assignable radio resources per 1UE in the radio access network) it is possible to identify. For example, each UE, based on the CSG ID or base station ID, when the base station constituting the radio access network is a femto base station is larger allocatable radio resources per 1UE in the radio access network it is determined that the.
Further, the signaling generator 610 of the UE100 creates a candidate codec mode set in consideration of the possibility of all or part of the characteristics of the radio access network 408 (or 508) to the communication partner (UE 102) is connected. The signaling generating unit 610 of the UE100 describes a candidate codec mode set in the SDP offer.
An example of an SDP offer description shown in FIG. For example, if the CSG ID of the destination radio access network UE100 is "Alice-femto", as shown in FIG. 8, in the SDP offer, for example: it is described as "a = csg Alice-femto". The description method may also be another describing method is not limited thereto. Further, as a method to describe the set candidate of all or codec mode considering some of the potential characteristics of the destination radio access network of the communication partner (UE 102), for example, only numbers respectively associated with each codec mode set it may be described in the. This number, in FIG. 8 "0x00", is set to "0x01" and "0x02". Thus, it is possible to reduce the amount of description in the SDP. Further, not only the correspondence between the codec mode sets and numerical may represent a codec mode set by a symbol, or other methods.
For example, numerical values ​​shown in FIG. 8 (number) "0x00", "0x01", and "0x02", showing the correspondence between the meanings and codec mode set that corresponds to those represented by the these numerical values ​​in FIG.
Specifically, it is shown in FIG. 9 "0x00" means, for example, the normal communication is a communication that is different from the communication corresponding to the later-described "0x01" and "0x02". Then, shown in FIG. 9 "0x00", the bit rate (6 kbps longitudinal, longitudinal 8 kbps, 12 kbps longitudinal), and corresponds to the codec mode set is low (Low) processing delay. In normal communications, for example, to initiate a call at a low bit rate among the bit rates that can be selected, then selects the codec bit rate depending on the quality of the communication path between UE. "0x01" is shown in FIG. 9, a communication (HeNB + IP core network communication. For example, referring to FIG. 5 (private home)) having passed through the IP core network means, bit rate (6 kbps longitudinal, longitudinal 8 kbps, before and after 12 kbps, a 24kbps so), it corresponds to the codec mode set is low (low) processing delay. Further, "0x02" is communication via the Local GW (HeNB + local network communication. For example, referring to FIG. 4 (enterprise telephone network)) means a bit rate (6 kbps longitudinal, longitudinal 8 kbps, before and after 12 kbps, 24 kbps is around), corresponds to the codec mode set is high (high) processing delay.
That is, in FIG. 9, "0x02" codec mode set that gives a good codec Most quality (high bit rate, long time delay) is followed by "0x01", in the order of "0x00", the codec mode set with less quality (low bit rate, low latency reduction) becomes. Mapping shown in FIG. 9 may be stored in advance in the terminal (UE100,102).
Generated by signaling generator 610 of the UE100 signaling message (SIP INVITE message) via the wireless transmission section 602, and transmitted to the UE102 through IMS network 126.
Meanwhile, radio receiving section 600 of the UE102 receives a signaling message sent from the UE 100 (SIP INVITE message).
Destination determination unit 604 of the UE102, like the UE 100, to determine the characteristics of the radio access network 408 (or 508) which UE102 is connected.
Information comparing unit 606 of the UE102 is a signaling message received, information about the characteristics of the destination radio access network of the communication partner (UE 100) (radio access network 406 (or 506)) to confirm whether it is written. Information comparing unit 606, if the information about the characteristics of the radio access network is described, comparing the characteristics of the radio access network of a communication partner, and a characteristic of its own the connection target radio access network. Then, the information comparing unit 606 determines the communication mode between the host machine (UE 102) and the communication partner (UE 100) (any communication is possible or). For example, the information comparing unit 606, a CSG ID transmitted as the information about the characteristics of the destination radio access network of the communication partner (UE 100), CSG ID is information about the characteristics of the destination radio access network of its own (UE 102) to compare the door. Then, the information comparing unit 606, the own device and whether a communication partner is connected to the HeNB in ​​the same enterprise telephone network, whether they are connected to the HeNB of private home with each other, or, either or both There what is connected to the public base station (eNB), it is determined. Or, the information comparing unit 606 compares the information about the characteristics of the destination radio access network of the communication partner (UE 100) and the own apparatus (UE 102), whether both physically located near (e.g., data number of hops of the route may determine whether) or less than a predetermined threshold value.
Mode determination unit 608 of the UE102, based on the determination result of the information comparison unit 606 (communication form between the own apparatus as a communication partner), codec mode set (bit rate, processing delay, the combination of the number of channels, etc.) decide.
An example of a process in the information comparing unit 606 and the mode determining section 608 in the embodiment shown in FIG. 10. Figure 10 is a flowchart showing a flow of processing in the information comparing unit 606 and the mode determining unit 608.
10, in ST 101, the information comparing unit 606, the destination of its own to determine whether the HeNB. If the destination of its own is not the HeNB (ST101: NO), the ST 102, mode determination section 608 selects the codec mode set for normal communication (0x00).
If the destination of its own is HeNB (ST101: YES), the ST 103, the information comparing unit 606, the connection destination of the communication partner to determine whether the HeNB. If the destination of the communication partner is not the HeNB (ST103: NO), the ST 104, mode determination section 608 selects the codec mode set that corresponds to the normal communication (0x00). That is, the mode determining unit 608, if not the HeNB in ​​one of the connection destination of its own and the communication partner, selects the codec mode set that corresponds to the normal communication (0x00).
If the destination of the communication partner is a HeNB (ST103: YES), the ST105, information comparing unit 606, the mutual connection of the own machine and the communication partner is determined whether or not the same local network.
If the mutual connection of the own machine and the communication partner is not within the same local network (ST105: NO), the ST 106, mode determination section 608 selects the codec mode set that corresponds to the "HeNB + IP core network communication (0x01)" . If the mutual connection of the own machine and the communication partner is in the same local network (ST105: YES), the ST 107, mode determination section 608 selects the codec mode set that corresponds to the "HeNB + local network communication (0x02)" to.
That is, the mode determining unit 608, a communication form between the own apparatus and the communication counterpart selects the highest quality codec mode set in the case of "HeNB + local network communication". The mode determination unit 608, the communication mode is the case of "HeNB + IP core network communication", to select the high-quality codec mode set than in "normal communication".
Thus, the mode determination unit 608, based on the information comparison unit 606 by the own apparatus and the good whether the communication path quality can be secured with the communication partner determination result, the own apparatus and the communication partner determining codec mode (or codec mode set) to be used for communication between. That is, when the mode determining unit 608, it is determined that can be secured (e.g., ST 101 shown in FIG. 10: YES and ST 103: YES), the codec mode to be used (or codec mode set), High Quality (high bit and a long selecting a codec mode set is the delay) ( "0x01" or "0x02"). On the other hand, the mode determining unit 608, when it is determined that it is unknown whether it secured (e.g., shown in FIG. 10 ST 101: NO or ST 103: NO), the codec mode to be used (or codec mode set), usually codec mode set (i.e., described above, any communicable codec mode set in the communication status) selects.
Signaling generating unit 610 of the UE102 describes a codec mode set that was selected by the mode determining unit 608, a signaling message (e.g., SIP 183 Session Progress message) for call control part of, for example, the SDP answer. In this case, signaling generator 610, a CSG ID or base station ID of the destination HeNB402 (or 502) of the UE 102, it may be described in the SDP answer or SIP header.
An example of a SDP answer the description shown in FIG. 11. For example, CSG ID of the destination radio access network UE100 is that shown in FIG. 8 "Alice-femto", CSG ID of the destination radio access network UE102 is assumed to be "Bob-femto" (ST101 in FIG. 10 , ST103: YES). Moreover, CSG ID ( "Bob-femto") and radio access network of a radio access network and the CSG ID ( "Alice-femto") is not in the same local network (ST105 in FIG. 10: NO).
In this case, the information comparing unit 606 of the UE102, for example, determines that the own apparatus and the communication partner is connected to the HeNB which is installed private home or the like to each other, the mode determination unit 608, "HeNB + IP core network communications ( to select the codec mode set corresponding to 0x01) ". Therefore, the SDP answer as shown in FIG. 11, "0x01" is described as a codec mode set that was selected. Incidentally, as shown in FIG. 11, it may be described information about the characteristics of the destination radio access network UE102 to (Bob-femto) in the SDP answer.
Generated by signaling generator 610 of the UE102 signaling message (e.g., SIP 183 Session Progress message) is sent to the UE100 through the IMS network 126 via the radio transmission section 602.
Mode determination unit 608 of the UE100 through the wireless reception unit 600 receives a signaling message sent from the UE 102, based on the contents described in the SDP answer, it determines the codec mode set. In an example of FIG. 11, the mode determination unit 608 of the UE100 determines the codec mode set that corresponds to the "HeNB + IP core network communication (0x01)". In this case, the CSG ID or base station ID of the destination UE100 communication partner (UE 102), may be used as reference information.
Thus, in the present embodiment, at the time of starting communication between the UE, the calling UE may communicate CSG ID (or base station ID) as information about the characteristics of the destination radio access network of its own party notifies the counterpart terminal (the called UE) is. The called UE may notified from the partner terminal (calling UE) is communicating, the information about the characteristics of the destination radio access network of the counterpart terminal (caller) (CSG ID, etc.), own apparatus comparing the information about the characteristics of the connection target radio access network (CSG ID, etc.). Then, UE of the called determines communication form between the own device and the destination terminal. The called UE's, if it can be determined that can be reliably ensure good quality communication path between UE, as codec mode set at the start of communication, a higher quality and a long delay codec mode (or codec mode set) to select. This, for example, the communication form is communication within the telephone network company, or a case where the communication in a private home.
Accordingly, each UE from the start of communication (initial stage of communication), appropriate codec mode set in accordance with the quality of the communication path (bit rate, processing delay, the combination of the number of channels, etc.) can be selected.
In this embodiment, the present invention has been described an example of applying to the selection of the codec mode sets is not limited thereto. Characteristic determining the connection target radio access network according to the present invention, other negotiation required for communication, for example, (the number of frames sent at a time, RTP payload format, etc.) and the transfer format of communication data used in negotiating the transmission and reception it may be applied to the format selection.
Further, in the present embodiment, it is only necessary to notify the information each UE owns (information about the characteristics of the radio access network) to a communication partner, the radio access network in either the network side and the UE side situation ( there is no need to monitor the quality) of a communication path. Therefore, it is possible to suppress the load on the network and UE for the selection of an appropriate codec mode (or codec mode set).
Therefore, according to this embodiment is used as negotiation information the information terminal is held. Thus, without applying both the processing load associated with the beacon or signaling transmission and reception in the network and the terminal (UE), to determine the status of the communication path between the terminal selects an appropriate codec mode (or codec mode set) be able to.
In this embodiment, UE-written in a part of the signaling message in the first embodiment, information about the characteristics of the radio access network is confirmed by the network node of the IMS network. The network node operator to provide a network, that is, the network node having the processing functions for call control.
Figure 12 is a block diagram showing a configuration of a network node according to the present embodiment.
At the network node shown in FIG. 12, the receiving unit 700 receives the signaling message from the UE, and outputs a signaling message to the signaling analyzer 704.
Transmitting unit 702, a signaling message received from the signaling generation part 708 is transmitted to the destination UE signaling messages.
Signaling analysis unit 704 analyzes the information described in the signaling messages received from the receiving unit 700, the necessary information is confirmed, and outputs the appropriate confirmation unit. For example, the signaling analyzer 704, among the information described in the signaling message, information about the characteristics of the radio access network that the UE is currently connected (e.g., CSG ID or base station ID, etc.), the terminal position information confirming unit and passes to 706.
Terminal position information confirming unit 706, information about the characteristics of the radio access network the UE that sent the signaling message is currently connected (CSG ID or the like. That is, the position information of the terminal) whether it is appropriate (validity) to make sure. In this case, the terminal position information confirming unit 706 may perform the check by using the information held in the network node, another network node (e.g., MME (Mobility Management Entity) and HSS (Home Subscriber Server) etc.. "3GPP TS 23.002 v10.0.0," Network Architecture "," a reference) and may be carried out to confirm in cooperation. Terminal position information confirming unit 706 outputs the confirmation result to the signaling generator 708.
Signaling generating unit 708, overwriting the description that need to be modified in a signaling message sent from the UE. For example, in the terminal position information confirming unit 706, and information about the characteristics of the radio access network is determined to be valid. In this case, the signaling generator 708, information indicating the validity confirmed by a network entity, flags, numbers, and may be added to the signaling messages described how the text or the like. Further, in the terminal position information confirming unit 706, when it is determined that the information about the not appropriate characteristics of the wireless access network, signaling generating unit 708, from the signaling message, may delete the information about the characteristics of the radio access network . Or, signaling generating unit 708, information (validity of check result) indicating the confirmation result at the terminal position information confirming unit 706 may add to the signaling message. Modified signaling message via the transmission unit 702, is sent to the destination of the signaling message.
UE which has received the signaling message, for example, to identify that the validation already by a network entity. In this case, UE, in the same manner as the first embodiment, by using the information about the characteristics of the destination radio access network of the communication partner described in the signaling message, determines the communication mode of the communication partner. Meanwhile, UE which has received the signaling message, for example, and can not be identified to be a validation completion by the network entity. In this case, UE used the information about the characteristics of the destination radio access network of the communication partner does not perform determination of the communication mode of the communication partner. That, UE, based on validity of the confirmation result at the network node may be determined whether to determine the form of communication with the partner terminal.
Thus, UE is able to accurately determine the form of communication between the own apparatus and the counterpart terminal. That is, it is possible to improve the reliability of communication style determining between the own apparatus and the counterpart terminal. Therefore, according to this embodiment, as compared with the first embodiment, the status of the communication path between terminals can be determined more accurately, can select an appropriate codec mode (or codec mode set).
In this embodiment, the network node of the IMS network entity may be a (P-CSCF and S-CSCF, and the like. "3GPP TS 23.002 v10.0.0," Network Architecture "" See), within a company IP-PBX may be (Internet Protocol private Branch eXchange) or the like used in constructing the dedicated telephone network. For example, which is one P-CSCF of the IMS network entities (P-CSCF108,116 shown in FIG. 4 or FIG. 5), check the contents of the SDP, the function of rewriting is provided. Therefore, the use of P-CSCF as the network node according to the present embodiment, it is possible to utilize also the confirmation information to confirm functionality of SDP on the characteristics of the radio access network, without increasing the network side load, the present embodiment it is possible to obtain the same effect as in.
Further, in the present embodiment, the information that network nodes the validity confirmed been described to notify UE, information indicating the validity confirmed may not be notified to the UE. For example, as discussed above, the network node deletes the information about the characteristics of the radio access network it is determined that it is not appropriate from the signaling message. In this case, the destination UE signaling message, it is unnecessary to perform communication form of determination using the incorrect information. Therefore, even if it does not signal the information network nodes the validity confirmed the UE, as in this embodiment, it is possible to improve the reliability of the communication mode determination between terminals. For example, the network node, the communication of the UE each other connected to the IP core network of the same operator may not notice the information indicating the validity confirmed. Then, instead, the network node, the communication of the UE each other are connected to the IP core network of different operators may notify the information indicating the validity confirmed.
In this embodiment, in addition to information about the characteristics of the connection destination access network of the UE, using the information indicating the presence or absence of potential UE handover, codec mode (or codec mode set) selection is made. Further, in this embodiment, when a change in the presence or absence of a handover possibilities on the way the call occurs, change of codec mode (or codec mode set) is performed.
Figure 13 is a block diagram showing the configuration of the terminal (UE100,102) according to the present embodiment. Incidentally, in FIG. 13, a description thereof will be given the same reference numerals with the same configuration as FIG. Terminal shown in FIG. 13, in addition to the configuration of the terminal shown in FIG. 7, the state analysis unit 812 and the mode change unit 814 is added.
In the terminal shown in FIG. 13, the state analysis unit 812, the type and condition of the own device, analyzing whether there is a possibility of handover in its own device (decision) to. For example, if the UE is a stationary terminal for a telephone conference system, the state analysis unit 812, the possibility of its own handover is judged to be low. Further, the own device is stationary even a small UE device, if the state of being connected to a speaker and a microphone or the like, the state analysis unit 812, the possibility of its own handover is judged to be low. Alternatively, provided the ship telephone conference mode setting unit (not shown), (the operation of the user interface of the UE) explicitly by the user if the conference call mode is selected, the state analysis unit 812, the own device It is judged to be low likelihood of the handover. State analyzer 812, the information indicating the analysis result of the start of communication, and outputs to the signaling generator 610. Thus, the analysis result (information indicating the presence or absence of possibility of its own handover) is added to the signaling message is notified to the communication partner.
The state analysis unit 812 analyzes the state of its own in the middle communication, if in the course communication state of the own apparatus is changed, and outputs the changing contents of the state of its own to the mode change unit 814. For example, stationary equipment, UE communicating is connected to a speaker and a microphone and the like, when it becomes these devices disconnected, the state analysis unit 812 in the multifunction peripheral and the these devices not connected it became outputs the mode change unit 814 a.
Mode change unit 814, based on the information input from the state analysis unit 812, changes the codec mode currently in use (or codec mode set). Incidentally, codec mode (or codec mode set) the requested change to the communication partner may be carried out by sending a signaling message for the call control. Furthermore, such request indicates the header of the communication data (e.g. RTP header, the header portion of the RTP payload, the control information of the communication data, or, RTCP (RTP Control Protocol)) changes to the codec mode (or codec mode set) it may be carried out by adding the change information.
Next, with reference to FIG. 13, the processing in the terminal (UE 100 and UE 102) according to the present embodiment will be described in detail.
In the following description, similarly to the first embodiment, in FIG. 4 or FIG. 5, current, UE 100 is connected to HeNB400 or HeNB500, UE 102 is connected to HeNB402 or 502, make a call with respect to UE 102 from the UE 100 to.
State analyzer 812 of the UE100 from the type and condition of the own device, analyzing whether there is a possibility of handover in its own device (decision) to.
Signaling generating unit 610 of the UE100 describes information showing an analysis result, some signaling messages for call control, for example, the SDP offer (or SIP header). Further, the signaling generator 610, as in the first embodiment, to create a candidate communication partner (UE 102) to connect to a wireless access network characteristics codec mode considering the (or codec mode set). Further, the signaling generator 610, as in the first embodiment, the communication partner (UE 102) all of the information about the UE 102 possibilities such as a handover or codec mode considering some of the possibilities (or codec mode set) to create a candidate. Signaling message including these information (SIP INVITE message) is sent to the communication partner (UE 102) via the radio transmission section 602.
An example of an SDP offer description shown in FIG. 14. For example, if it is determined that there is a low possibility of handover UE 100 in the state analysis unit 812 of the UE 100, as shown in FIG. 14, in the SDP offer, for example, "a = handover: low" and is described. Instead it described as whether the "low" or "High" of the potential handover, may be indicated by numerical values ​​or symbols or the like. For example "0x00" was the "possibility of handover high", the "0x01" in case of the "possibility of the handover lower" is handover possibility is low state, "a = handover: 0x01" and is described that. Description method is not limited thereto, it may be another description method.
Then, in this embodiment, the codec mode set (including codec mode set when there is a low possibility of a UE handover), shown in Figure 15 the correspondence between the numerical value (number) indicating the codec mode set. In Figure 15, the correspondence relationship shown in FIG. 9 ( "0x00", "0x01", "0x02") was added, "0x03" is set. FIG 15 "0x03" means a telephone conference communication, the bit rate is around 100kbps, corresponds to the codec mode set is high (High) processing delay.
On the other hand, UE 102 information comparison section 606 of the (called party) is and the destination information about the characteristics of the radio access network, of its own terminal status (UE 102 status (analysis result of the condition analysis unit 812 of the UE 100) a terminal condition comparing the information about the characteristics of the analysis result) and the destination wireless access network at the analysis unit 812. The former is the information contained in the signaling message sent from UE 100 (SIP INVITE message). Specifically, the information comparing unit 606, based on the comparison result, determines the communication mode between the host machine (UE 102) and the communication partner (UE 100) (any communication is possible or). For example, the information comparing unit 606, UE100,102 are both connected to the HeNB, and, when there is no possibility of the handover is determined to be communication such as a fixed telephone (high quality communication).
An example of a process in the information comparing unit 606 and the mode determining section 608 in the embodiment shown in FIG. 16. Figure 16 is a flowchart showing a flow of processing in the information comparing unit 606 and the mode determining unit 608. Incidentally, in FIG. 16, description thereof will be denoted by the same reference numerals perform the same processing as FIG. 10.
16, mutual connection destination of its own (UE 102) and the communication partner (UE 100) is the same local network (ST105: YES). In this case, the information comparing unit 606, in ST 201, it is determined whether there is a possibility of its own handover, it determines whether there is a possibility of handover of the communication partner in ST 202. If there is a possibility of its own handover (ST201: YES), or if there is a possibility of handover of the communication partner (ST 202: YES), the ST 107, mode determination section 608, "HeNB + local network communication (0x02 ) to select the codec mode set that corresponds to the ".
On the other hand, there is no possibility of its own handover (ST201: NO), and when there is no possibility of the handover of the communication partner (ST202: NO), the ST 203, mode determination section 608, "conference call communications (0x03 ) to select the codec mode set that corresponds to the ".
Both this way, mode determination section 608 determines whether or not is determined by the information comparison unit 606 can ensure good quality communication path between the own apparatus and the communication counterpart, and own apparatus and the communication partner in on the basis of whether there is no possibility of the handover, selects the codec mode set to be used for communication between the own apparatus and the communication partner. More specifically, it is determined that can be secured communication path good quality (for example, ST 101 shown in FIG. 16: YES and ST 103: YES), and if there is no possibility of a handover in both (e.g., FIG. 16 to indicate ST 201: NO and ST 202: NO), among the selectable codec mode set, selects the codec mode set (0x03) is the most high-quality and long delays.
Signaling generating unit 610 of the UE102 describes a codec mode set that was selected by the mode determining unit 608, a signaling message (e.g., SIP 183 Session Progress message) for call control part of, for example, the SDP answer. In this case, signaling generator 610, CSG ID or base station ID of the destination HeNB402 (or 502) of the UE 102, and the presence or absence of a handover possibility of UE 102, may be described in the SDP answer or SIP header.
An example of a SDP answer the description shown in FIG. 17. For example, a CSG ID of the destination radio access network UE100 is shown in FIG. 14 "Panasonic-femto", CSG ID of the destination radio access network UE102 is assumed to be "Panasonic-femto" (ST101 in FIG. 16 , ST103: YES). Also, the a radio access network of a radio access network and CSG ID of the CSG ID ( "Panasonic-femto") ( "Panasonic-femto") is the same local network (ST105 in FIG. 16: YES). Furthermore, UE 100 and the possibility of both handover UE102 is low (ST 201 in FIG. 16, ST202: NO). In this case, the information comparison unit 606 of the UE 102, is connected to the HeNB local network UE100 and UE 102 is within the same enterprise each other, and determines that there is no possibility of the handover are both UE100 and UE 102. The mode determination unit 608 selects the codec mode set that corresponds to the "conference call communications (0x03)". Therefore, the SDP answer as shown in FIG. 17, "0x03" is described as a codec mode set that was selected.
Incidentally, as shown in FIG. 17, the connection destination information about the characteristics of the radio access network (Panasonic-femto) of the UE 102, and the presence or absence of the possibility of handover UE102 (handover: low) may be described to SDP answer .
Mode determination unit 608 of the UE100 through the wireless reception unit 600 receives a signaling message sent from the UE 102, based on the contents described in the SDP answer, it determines the codec mode set. In an example of FIG. 17, the mode determination unit 608 of the UE100 determines the codec mode set that corresponds to the "conference call communications (0x03)". In this case, CSG ID or base station ID of the destination UE100 communication partner (UE 102), or the presence or absence of possibility of handover, may be used as reference information.
Also, if the state of the communication middle UE100 or UE102 is changed, the mode changing unit 814 may perform changes of the codec mode set. For example, the current codec mode set and corresponds to the "telephone conference communication (0x03)". In this case, the mode changing unit 814, the header portion of the RTP payload or a codec mode request field of RTCP feedback describes a codec mode set after the change (e.g., "HeNB under + local network communication (0x02)" (e.g., see FIG. 18). Then, the mode changing unit 814, at the same time urge the change to the communication partner, ship codec mode (or codec mode set) also changes. Further, instead of the change destination codec mode set, change destination bit rate, delay mode, the individual codec modes number of channels or the like may be specified directly to the codec mode request field. Further, the change destination bit rate, delay mode, number of channels, etc. plurality of codec mode (or codec mode set ) also across the .
Thus, the UE, not only characteristics of the radio access network, by using the presence or absence of potential handover, as compared with the first embodiment, the communication mode between the correspondent terminal as a communication partner with its own device it is possible to more finely determined. Therefore, it is possible in UE, in response to the determined said communication mode, selecting a higher quality codec mode (or codec mode set).
Furthermore, UE is in the same apparatus or counterpart terminal in communication, when the change in potential handover to change the codec mode currently in use (or codec mode set). That, UE, even during communication, when one of the own apparatus or the communication partner is changed, that is, if the communication form between the own apparatus and the communication partner may change, the change depending on the, to change the codec mode (or codec mode set). Thus, the UE, it is possible to use the appropriate codec mode (or codec mode set) in accordance with a change in the communication form in during communication.
Therefore, according to this embodiment, as compared with the first embodiment, the status of the communication path between terminals can be made finer determination can select a more appropriate codec mode (or codec mode set).
In this embodiment, as in the codec format mentioned in the example of the first to third embodiments, is recommended for the codec mode type often codec format such as bit rate, an example of RTP payload format less to show. Incidentally, RTP payload format example of the present embodiment is not intended to the selection of the first to third embodiments negotiation process or codec mode described in (or codec mode set) is limited when done.
Here, one method of the codec (or mode), or, as the bit rate in a plurality of methods (or modes) that can be used to switch from one another in a single real-time session such as a telephone call, system X (if multi-rate and codec) in 12,24kbps is assumed to be supported. Then, the system Y (assumed as scalable codec), as such a bit rate, it is assumed that the basic unit 8,12,24Kbps, first extension 4,12Kbps, the second extension portion 8kbps supported. Here scheme X, type Y may be the different codecs, may be different modes of medium of a single codec.
Here, the first extension portion and the second extension portion of a scalable codec, such as Super Wideband extension and Full may be Band extension, or may be a Enhance corresponding extension or stereo extension like. Also, methods both of X and type Y, and having both of their respective a delay mode different delay A and the delay B, or either one. Also, two of the number of channels one channel (mono) and 2-channel (stereo), or to support only one channel (mono).
Figure 19 is a first example showing the number correspondence (Frame Type Index) and RTP Payload format for codec mode (bit rate and the combination in the present embodiment) of the present embodiment.
As shown in FIG. 19A, different Frame Type Index for all combinations of bit rates are assigned respectively. Figure 19B is an example of an RTP payload format in the first embodiment. As shown in FIG. 19B, in addition to the data portion of the RTP payload, the header portion of the RTP payload, as information of the RTP payload, Frame Type Index and Q bits are added. Here, the data portion of the RTP payload includes padding bits (padding bit) to be added to adjust the encoded data and the RTP payload length 1frame minute. Frame Type Index included in the header portion of the RTP payload, indicating the combination of Figure 19A throat is used. Q bits contained in the header portion of the RTP payload is equivalent to Q bit described in Non-Patent Document 1, an Indicator of Frame Quality.
For the delay mode, be included in the header portion of the RTP payload may not be included. If the inclusion of a delay mode in the header portion of the RTP payload may be assigned a Frame Type Index moistened with delay mode in the table of FIG. 19A. Then, bits representing the delay mode in the RTP payload format (e.g. delay A is "0", the delay B is "1") may be included.
As shown in FIG. 19B, the RTP payload format in the first embodiment, no fields such as Codec Mode Request that is described in Non-patent Document 1. It is possible to suppress the size of the RTP payload by not have this field. However, if there is no problem with the size of the RTP payload (that is, if there is no problem with adding a few bits in the RTP payload format) may include fields such as Codec Mode Request. In the case where no field Codec Mode Request, when determining the change of codec mode in the communication partner (or codec mode set), for example, may be used RTCP.
Furthermore, (in the single IP packet) a plurality of frames (encoded data for one frame) may be simultaneously sent. In this case, among the plurality of frames being sent simultaneously, the bit for which frame to identify whether the last frame may be included in the header portion of the RTP payload.
The bits and fields included in the header is not limited to those listed in this example.
Further, in this example, provided with a series of Frame Type Index to the combination of the bit rate, or bit rate of the system X and type Y as shown in FIG. 19A, is the same codec mode as if method X and type Y is It gave the example that can be handled in Kano as. However, with respect to method X and type Y, each given separately Frame Type Index and RTP payload formats, or may be used independently.
Figure 20 is a second example showing the number correspondence (Frame Type Index) and RTP Payload format codec mode in the present embodiment (the bit rate in this embodiment).
As shown in FIG. 20A, Frame Type Index is assigned respectively only the bit rate. Incidentally, Frame Type Index shown in Figure 20A specifies only the bit rate, it will not be described which codec or a combination of the codec is used. Such a combination, of the combination of FIG. 19A (the first example), a codec mode that applies to each of the bit rate is applicable for all. For example, in FIG. 20A, Frame Type Index value 1 (bit rate: 12 kbps) and which was a. In this case, the content of the frame, Frame Type Index "0" corresponding to the codec bit rate in FIG. 19A is 12 kbps (12 kbps scheme X), "3" (12 kbps scheme Y) or "5" ( the possibility of a combination of basic unit 8 kbps, and the first extension portion 4 kbps) in a manner Y. However, when the FIG. 20A corresponds only to the codec scheme Y is, (12 kbps scheme Y) Frame Type Index "3", or, Frame Type Index "5" (Method Y in the basic unit 8 kbps, and the the two possibilities one extension 4 kbps). Among combinations of these codec format and bit rate, or choose which combination encoder is determined. Information indicating the combination of the encoder is determined, it may be included in the encoded data.
Figure 20B is an example of an RTP payload format in the second embodiment. As shown in FIG. 20B, in addition to the data portion of the RTP payload, the header portion of the RTP payload, as information of the RTP payload, Frame Type Index, Q bits include Codec Mode Request and R bits. Data portion of the RTP payload includes padding bits (padding bit) to be added to adjust the encoded data and the RTP payload length 1frame minute. Frame Type Index of the header portion of the RTP payload indicates whether being used the bit rate of the Figure 19A throat. Q bit is equivalent to Q bit described in Non-Patent Document 1.
Codec Mode Request is a field to request a change of codec mode in the communication partner (bit rate), the Frame Type Index in Figure 20B is used. Encoder of a communication partner received a Codec Mode Request encodes select optimal among combinations of codecs that provide the specified bit rate. For example, if the value of the Frame Type Index shown in Codec Mode Request was 1 (12 kbps), the contents of the frame, Frame Type Index "0" in FIG. 19A, the combination of "3" or "5" there is a possibility. However, when the FIG. 20A corresponds only to the codec scheme Y is, (12 kbps scheme Y) Frame Type Index "3", or "5" (the basic unit in a manner Y 8 kbps, and a first extension portion the possibility of the two ways of 4kbps). The encoder performs encoding using a combination it is determined that the optimum of these.
R bits shown in FIG. 20B is a field indicating a request which can not be requested to the other party in the Codec Mode Request. For example, R bits specify the number of channels (e.g., mono = "0", the stereo = "1") or the specified delay (e.g., delay A = "0", the delay B = "1") indicating the like. Since R bits indicating both delay for example, the number of channels and may be two or more bits. Note that this R bit, if not switch the delay or the number of channels in communication may not be present in the header portion of the RTP payload.
As for the delay mode may be assigned a Frame Type Index in the delay mode is also included in the table of FIG. 20A. For example, measuring the delay A for the bit rate assigned to up to "8" from the Frame Type Index "0". Furthermore, it extended to 31 the value of the Frame Type Index shown in Figure 20A, Frame Type Index "11" (not shown) measuring delay B with respect to the bit rate assigned to up to "19" from. When it is not included delay mode in the table of FIG. 20A may be included a bit indicating a delay mode in the header portion of the RTP payload relative to the encoding side (their side). Also, in terms of the number of channels of the encoding end (his side), as in the delay mode, (by extending the value of the Frame Type Index) in the table of FIG. 20A may be included. Further, it may be included a bit indicating the specified number of channels in the header portion of the RTP payload may be included in the encoded data itself.
As in the second example, by specifying only the bit rate of the codec, it is possible to perform optimum encoding the encoding side, and, Frame Type Index and Codec Mode Request bit in the header portion of the RTP payload it is possible to reduce the number. Further, by including a Codec Mode Request header portion of the RTP payload may be transmitted to the communication partner, a change request of the codec mode (or codec mode set) rapidly. However, if the request for rapid codec mode (or codec mode set) changes to the communication partner is not required, if Codec Mode Request and R bits is required, without including them in the header portion of the RTP payload, the RTCP it may be notified to the other party, including. Furthermore, (in the single IP packet) a plurality of frames (encoded data for one frame) may be simultaneously sent. In this case, among the plurality of frames being sent simultaneously, the bit for which frame to identify whether the last frame may be included in the header portion of the RTP payload. The bit and the field included in the header portion of the RTP payload is not limited to those listed in this example.
Figure 21 is a third example showing the number correspondence (Frame Type Index) and RTP Payload format codec mode in the present embodiment (the bit rate and the type in this example).
As shown in FIG. 21A, a multi-rate codec, the basic unit of the scalable codec, and are assigned different Frame Type Index in each of the extension portions of the scalable codec.
Figure 21B is an example of an RTP payload format in the third embodiment.
In a third example, as shown in FIG. 21, the layer in scalable codec (basic portion, the first extension, the second extension portion) header portion of the RTP payload is attached to each. In Figure 21B, C bits contained in the header portion of the RTP payload indicates whether extension of subsequent exists. Thus, the basic part and the extension part may be sent as a separate IP packet. Thus, if the network or wireless access section, etc. is congested, there is an advantage discard IP packets having a low priority extension.
Incidentally, the header of each RTP payload, or only the header portion of the RTP payload of the base portion, may be added to Codec Mode Request and Q bits. For the method of changing requirements of the codec mode for the description and the communication partner of the delay mode (or codec mode set) is the same as in the first embodiment (FIG. 19) and the second example (FIG. 20). Furthermore, (in the single IP packet) a plurality of frames may be simultaneously sent. In this case, among the plurality of frames being sent simultaneously, the bit for which frame to identify whether the last frame may be included in the header portion of the RTP payload. The bit and the field included in the header portion of the RTP payload is not limited to those listed in this example.
Further, in this example gives, each bit rate mode, as shown in FIG. 21A X (e.g. multi-rate codec) and method Y (e.g. scalable codec), or for the type of bit rates, a series of Frame Type Index Example the mentioned. That is, an example that can be handled as if method X and type Y are the same codec mode as if. However, with respect to method X and type Y, each given separately Frame Type Index and payload format, or may be used independently.
In this embodiment, an example that contains Q bits, Q bits may be omitted.
In this embodiment, when the RTP payload length when limited to a plurality of fixed length, and, the combination representing a single bit rate (or bit rate of the peripheral thereof) may more present, preferred RTP Payload Format and its processing method. This is the case, for example, a method Y described in the fourth embodiment.
As an example of RTP payload length is limited to a plurality of fixed length, as in the radio section of the mobile phone network, there are cases where the transport block size is determined.
Figure 22 is a first example showing the number correspondence (Frame Type Index) and RTP Payload format codec mode in the present embodiment.
In a first example, (correspondence between Frame Type Index) number correspondence relative to gross bit rate (Gross Bit rate) is performed. Here, the gross bit rate, the RTP payload length kilobits per second: and those converted into (kbps kilo bit per second). Examples of numbers labeling of gross bit rate shown in FIG. 22A.
In the first example, a part of the information or information including the header portion of the RTP payload (e.g. Codec Mode Request field) may be selected by a call start negotiation (SDP offers and answers). At this time, the codec mode in the first and third embodiments (or codec mode set) (range of bit rates used, delay mode, etc.) when it is selected, the selected codec mode (or codec mode set) was together, the part of the information or information contained in the header portion of the RTP payload may be selected.
For example, "normal communication" or the like according to Figures 9 and 15, when the range of bit rates to be used is low, in order to reduce the proportion of the header portion of the RTP payload, the Codec Mode Request field of the RTP payload choose not included in the header portion is performed.
22B is negotiated at call start, or indicate when it is determined that not include pre Codec Mode Request field by the policy such as service operators in the header portion of the RTP payload, an example of RTP payload format. The Figure 22C, calls starting negotiation, or indicate when it is determined that inclusion of pre Codec Mode Request field by the policy such as service operators in the header portion of the RTP payload, an example of RTP payload format. In FIG. 22B and FIG. 22C, Q bits contained in the header portion of the RTP payload is equivalent to Q bit described in Non-Patent Document 1, an Indicator of Frame Quality.
The encoder side, the Frame Type Index, a bit rate corresponding to the length of the data portion of the RTP payload, which is determined from the length of the header portion of the RTP payload, combinations thereof, corresponding codec scheme bit is supported rates and their determined from among the combinations. Here, the length of the header portion of the RTP payload are those previously determined by a policy such as negotiation or service operator or the like during a call initiation. Information indicating the combination of the encoder is determined, it may be included in the encoded data. Incidentally, the policy such as service operator, when the change of the codec bit rate from the start of communication to the end is not performed, may not include a Frame Type Index in the header portion of the RTP payload. The constant bit rate in this case may be determined by negotiation at the call start, it may be predetermined by the policy such as service operators.
23, in the case where the information contained in the header portion of the RTP payload were determined through advance, number correspondence codec mode (Frame Type Index) and shows an example of a RTP payload format (second embodiment the example). That is to say, the first embodiment example, instead of the information contained in the header portion of the RTP payload as shown in (FIG. 22) is determined by negotiation at the start communication, the header portion of the RTP payload it is when the length had been predetermined.
In this example, as shown in FIG. 23A, the length of the data portion of the RTP payload kilobits per second (kbps: kilo bit per second) bit rate obtained by converting the relative, number correspondence (Frame association between the Type Index) is carried out. In this example, encoded data of 1frame content would be RTP packets per 20 ms. That is, 50 RTP packets are generated per second. The length of the header portion of the RTP payload, 7.2 kbps are gross bit rate shown in Figure 22A, 8 kbps, in the case of 9.6kbps and 5 bits, if the gross bit rate is 13.2 kbps, 16.4Kbps, the 24.4Kbps 8 are a bit. The number of RTP packets per second, the length of the header portion of the payload, and correspondence between the length of the gross bit rate and the header portion is not limited thereto.
FIG. 23B and FIG. 23C, if it contains Codec Mode Request field, and show an example of an RTP payload format may not include respectively. In this example, the information that may be included in the header portion of the RTP payload, and 4-bit Frame Type Index field, a 1 bit Q bit field and indicates an example where the 3-bit Codec Mode Request field.
The Frame Type Index field, enter the Frame Type Index corresponding in Figure 23A. Q bit field is equivalent to Q bit described in Non-Patent Document 1. Also, Codec Mode Request Of Frame Type Index in Figure 23A the field "0" and "5" (i.e., only the information necessary to request the bit rate changes to the communication partner) enters. Also, the Codec Mode Request field may be used to "6" or "7" as the value indicating that the bit rate changes to the encoder of the communication partner is not. Each field length or corresponding values ​​of the header portion of the RTP payload of this embodiment may be not limited thereto, the information contained in the header portion of the RTP payload of this example (field) is also not limited to this.
The first example of the present embodiment (FIG. 22), to specify the second embodiment (FIG. 23), the data portion of the RTP payload in accordance with the gross bit rate or gross bit rate by a length as Frame Type Index by doing so, optimum encoding can be performed in accordance with the length of the data portion of the RTP payload encoding side, and to reduce the number of bits of the Frame Type Index in the header portion of the RTP payload be able to.
Also it allows adjusting the length of the header portion of the RTP payload. By doing so, determine the length of the data portion of the RTP payload in accordance with such a range of the bit rate determined by the characteristics of the transmission path (codec mode of the first to third embodiments (or codec mode set)) and, it is possible to perform suitable encoding. In case you do not include a Codec Mode Request field in the header portion of the RTP payload, if change notification bit rate to the communication counterpart encoder is required, change notification by RTCP is performed.
In this embodiment, a method for use in a communication / call service operator policy codec mode (or codec mode set) a which provides a selection for each UE.
The operator policy, for example, (Method of codec) codecs used in preference, priority to the codec bit rate used, preferentially sampling rate used, the audio band used in preference (e.g. , NB, WB, SWB, FB), the priority and the processing delays used, and the number of channels and the like which are used in preference. As the operator policy, Besides, for example, whether to permit the bit rate or number of channel switching in the middle session, Source-Controlled Variable Bit Rate (VBR) operations (e.g., "3GPP2 C.S0052 -A Version 1.0 "Source-Controlled Variable-Rate Multimode Wideband Speech Codec (VMR-WB)" "to see) whether or not to permit, handover to the circuit switched network (for example," 3GPP TS23.216 v9.6.0 " Single Radio Voice Call Continuity (SRVCC); Stage 2 "" see) whether to permit, and, whether or not such to permit redundant frames and the like, such as described in non-Patent Document 2.
The UE, all or some of these operators policies before call initiation, i.e., acquired before the processing of ST11 shown in FIG. 2 is performed, it generates an SDP offer based on the acquired Poshiri.
As a method obtaining a policy (all or part), for example, signaling to the policy (all or part) sent from the operator of the network to the UE to include the, employs a method of obtaining from such signaling it may be.
This signaling can, for example, SIB (System Information Block. For example, "3GPP TS 36.331 v10.0.0," Evolved Universal Terrestrial Radio Access (E-UTRA) Radio Resource Control (RRC) Protocol specification "" see) even good. Also, this signaling can, for example, RRC (Radio Resource Control) connection establishment when signaling (e.g., "3GPP TS 36.331 v10.0.0," Evolved Universal Terrestrial Radio Access (E-UTRA) Radio Resource Control (RRC) Protocol specification "" it may be a RRC Connection Setup) described.
Moreover, policy, for example, sent with the IMS SIP signaling, such as the response message (200OK) (see, for example, "3GPP TS 23.228 v10.3.1," IP Multimedia Subsystem (IMS) Stage 2 "") for the Register it may be. Moreover, the policy, for example, may be included in the response message of a P-CSCF searches. If a policy is sent to the UE using the response message, a request from the UE a message (e.g. "3GPP TS 36.331 v10.0.0," Evolved Universal Terrestrial Radio Access (E-UTRA) Radio Resource Control (RRC) Protocol specification " RRC Connection Request according to "or indicating" in 3GPP TS 23.228 v10.3.1, "IP Multimedia Subsystem (IMS) Stage 2" Register according to "), is capable of receiving the response message operator policies parameters may be added.
As another method of obtaining the policy, UE, from the information about the policy that UE holds locally, on the basis of the information sent from the operator of the network to UE, an operator policy (all or a portion ) may be determined. For example, UE, the policy of each operator (all or part) holds locally, to obtain information about the operator (operator information) when connected to a certain operator's network, held in the local and the appropriate operator policy (all or part) from the database are selected. Operator information the UE acquires when connected to the operator's network, for example, it is obtained from the Cell ID (e.g., see "3GPP TS 23.003 v10.0.0," Numbering, addressing and identification (Release 10) "") .
As another method of obtaining the policy, UE is sent when connecting to the operator's network, QoS information (e.g. permitted for a call for each user, "3GPP TS 23.401 v10.2.1," Evolved Universal Terrestrial based on the Radio Access Network (E-UTRAN) access see (Release 10) ""), UE may select the codec bit rate and the like that are used in preference to a call.
Next, an example of a method of generating the SDP based on the policy acquired from the operator.
Figure 24 is policy of the operator (operator A) to be connected is calling UE (FIG. 24A), and shows the SDP offer generation example of calling UE (Fig. 24B). Further, FIG. 25 is a policy of the operator (operator B) the called UE is connected (Fig. 25A), and shows the SDP answer generation example of the called UE (Fig. 25B). Note that in FIG. 24 and FIG. 25, items that are not described in the policy or SDP, with respect to the value or the like, for example, the default of each method described in Non-Patent Document 2 is adapted. Also, how to write policies and SDP shown in FIG. 24 and FIG. 25 is an example and may be other description method.
For example, as shown in FIG. 24A, the policy of the operator A, "Codec: method 1, the audio band: SWB, bit rate: [12 kbps, 24 kbps] and VBR: (Allow VBR) permission" that policy (hereinafter, referred to as the policy A1) are included. Moreover, the policy of the operator A, "Codec: method 1, the audio band: WB, bit rate: [12 kbps, 24 kbps] and VBR: variable (Allow VBR)" that policy (hereinafter referred to as the policy A2) include It is. Furthermore, the policy of the operator A, "Codec: scheme 2, the audio band: NB, bit rate: [8 kbps, 12 kbps] and VBR: No (do not allow VBR)" that policy (called hereinafter a policy A3) is included.
Therefore, the calling UE, the operator policy to provide a communication or call service to the calling UE (here, the policy of the operator A shown in FIG. 24A) codec mode (or codec corresponding to at least a portion of the to create an SDP offer that contains the candidate of the mode set).
For example, as shown in FIG. 24B, the calling UE is part of a policy A1 shown in FIG. 24A, the codec: method 1, the audio band: SWB, bit rate: codec corresponding to {12, 24} kbps selecting a candidate of the mode set. Specifically, shown in FIG. 24B, the codec: method 1, the audio band: SWB, bit rate: [12 kbps, 24 kbps] a, VBR: Yes (Yes), or VBR: No (No) and a two codec mode set candidate (a = rtpmap: 97 and 98) is created.
Similarly, as shown in FIG. 24B, the calling UE is part of a policy A2 shown in FIG. 24A, the codec: method 1, the audio band: WB, bit rate: corresponding to {12, 24} kbps selecting a candidate of the codec mode set. Specifically, shown in FIG. 24B, the codec: method 1, the audio band: WB, bit rate: [12 kbps, 24 kbps] a, VBR: Yes (Yes), or VBR: No (No) and a two codec mode set candidate (a = rtpmap: 99 and 100) is created.
Similarly, as shown in FIG. 24B, the calling UE is an all policies A3 shown in FIG. 24A, the codec: scheme 2, the audio band: NB, bit rate: [8 kbps, 12 kbps] and VBR: No ( codec mode set of candidates corresponding to the disabled) (a = rtpmap: 101) to create a.
As shown in FIG. 24B, the candidate of the plurality of codec modes included in the SDP offer sent from the calling UE to the called-side UE (or codec mode set), communication or call to the calling side UE providing services include a candidate codec mode corresponding to at least a portion of the operator's policies (or codec mode set).
On the other hand, as shown in FIG. 25A, the policy of the operator B, "Codec: method 1, the audio band: WB, bit rate: 12 kbps and VBR: No (do not allow VBR)" that policy (hereinafter, a policy B1 call) are included. Moreover, the policy of the operator B, "Codec: scheme 2, the audio band: NB, bit rate: 12 kbps, and VBR: No (do not allow VBR)" that policy (hereinafter referred to as the policy B2) are included.
Therefore, the called UE, of the candidate codec mode set that is included in the SDP offer shown in Figure 24B notified from the calling UE, codec mode corresponding to at least part of the policy of the operator B shown in FIG. 25A from a set of candidates, and selects the codec mode set to be used to communicate with the calling UE.
Specifically, in FIG. 25B, the called UE a candidate codec mode set that corresponds to all policy B1 shown in FIG. 25A (Codec: Method 1, the audio band: WB, bit rate: 12 kbps, and, VBR : select a free (No)), as a codec mode set to be used to communicate with the calling UE.
Incidentally, codec mode (or codec mode set) selection method of the present embodiment may be used in conjunction with the codec mode (or codec mode set) selection method shown in the first to third embodiments.
In this way, according to this embodiment, calling UE be restricted according to the operator's policy to SDP offer codec mode (or codec mode set) described in the like the UE is connected it can. Furthermore, the called UE is a codec mode (or codec mode set) or the like is selected as the codec mode (or codec mode set) to be used to communicate with the calling UE to the operator policy to which the UE is connected it can be limited depending on. Thus, it is possible to avoid a complex SDP negotiation between the calling UE and the called UE. The operator can select a service (codec) based on the policy to the UE.
In this embodiment, the codec mode (or codec mode set) selection method will be described in the case of using an example different network configurations and network configuration used in the first to third embodiments.
Figure 26 is a network configuration according to the present embodiment, is a diagram showing an example of the positional relationship between the UE (terminal). Incidentally, in FIG. 26, UE 100, UE 102, IP core network 124 and the IMS network 126 that are denoted by the same reference numerals are the same as FIG.
AP2600 and AP2602 shown in FIG. 26 is an access point (Access Point), constituting the radio access network 2606 and a radio access network 2608, respectively. For example, AP2600 and 2602 may be an access point such as a Wi-Fi or WiMAX, may be a HeNB shown in FIG.
IP-PBX2604 is connected with the AP2600 and AP2602, for example, in tissue LAN such as corporations is a device for implementing the enterprise telephone network by IP telephone (network node having the processing functions for call control). For example, IP-PBX2604 the connection function with the operator of the IMS network, or have some of the functions of the operator's IMS network (e.g., "3GPP TS 22.809 v1.0.0, Feasibility Study on Support for 3GPP Voice Interworking with Enterprise see the IP- PBX (VINE) ""). Moreover, IP-PBX2604 is service present in the operator's IMS network 126 through an external IP network 2610 broadband network such as the Internet provider - is connected to the server 2612. Moreover, IP-PBX2604 has a function of terminating a gateway function or IMS signaling with the external IP network 2610 may have the combined (not shown) or the like.
Server 2612, for example, CSCF etc. S-CSCF100 shown in FIG. 1, AS (Application Server. For example, in "3GPP TS 23.228 v10.3.1," IP Multimedia Subsystem (IMS) Stage 2 "), or, both it may also be, or may be a device that connects with these CSCF or aS, and IP-PBX2604.
AP2614 is an access point connected to the IP core network 124. AP2614 can be a base station (eNB or HeNB). In addition, (see, for example, the "3GPP TS 23.402 v10.2.1," Architecture enhancements for non-3GPP accesses "") AP2614 is, may be the access point such as Wi-Fi or WiMAX.
In Figure 26, UE 100 is connected to the AP2600 in the radio access network 2606. In this case, for example, the destination determining unit 604 of the UE 100 (FIG. 7) is that of its own apparatus connection destination (wireless access network 2606) is determined to be a radio access network constituting a tissue LAN of a company or the like. In the case AP2600 is HeNB, connecting destination determination section 604, as in the first embodiment may determine a connection destination by using a CSG ID and the like. Also, if the AP2600 is an access point of a wireless LAN, such as Wi-Fi, connecting destination determination section 604, SSID (Service Set Identifier) ​​may determine a connection destination by using the like.
In this embodiment, UE 100 if (caller) makes a call to UE 102 (called side), as in the first to third embodiments, UE 100 is connected destination determination unit 604 (FIG. 7, FIG. 13 it may be selected to describe codec mode (or codec mode set) into the SDP offer according to the judgment of). It is also possible to select the codec mode (or codec mode set) in IP-PBX2604 without selecting codec mode (or codec mode set) with UE 100.
Figure 27 is a block diagram showing a configuration of IP-PBX2604 according to the present embodiment.
In IP-PBX2604 shown in FIG. 27, the receiving unit 2700, the calling UE (in this case UE 100) receives a signaling message from, and outputs a signaling message to the signaling intercept unit 2704.
Transmission unit 2702 transmits the signaling messages inputted from the signaling generation unit 2710, to the destination UE signaling message (the called UE. Here UE 102).
Signaling intercept unit 2704, even if the signaling messages inputted from the receiving unit 2700 is not addressed to the own apparatus (IP-PBX2604 IP address of), once intercept (intercept) the signaling message. Signaling intercept unit 2704 outputs the signaling messages intercepted in the signaling analyzer 2706.
Signaling analysis unit 2706 analyzes the information described in the entered signaling message, the information required to check, and outputs the appropriate confirmation unit. For example, the signaling analyzer 2706, to the called radio access network UE (UE 102) is currently connected to specify whether the subordinate IP-PBX2604, among the information written in a signaling message , the call destination information, passed to the terminal position information confirming unit 2708.
Terminal position information confirming unit 2708, whether the called UE (UE 102) is connected whether the radio access network that is currently connected is under the control of IP-PBX2604 (e.g., in the same enterprise telephone network ) to make sure. This confirmation is performed based on the calling destination information received from the signaling analyzer 2706. In this case, the terminal position information confirming unit 2708, IP-PBX2604 in using the registration information of UE102 of may be performed to confirm that all of the registration information of UE102 present in the IMS network 126 through the server 2612 or one it may be carried out to confirm using parts. Or, the terminal position information confirming unit 2708 may perform the check by using both of these information. Terminal position information confirming unit 2708 outputs the check result to the signaling generator 2710.
Signaling generating unit 2710, overwriting or create new description that need to be modified in the transmitted signaling message from a calling UE (UE 100). For example, in the terminal position information confirming unit 2708, and the called UE (UE 102) is determined to be the same enterprise telephone network (here under IP-PBX2604) present in the. In this case, the signaling generator 2710, the description contents of SDP offers signaling message, newly written or overwritten to contain the same codec mode (or codec mode set) used for the call in the telephone network companies. In this case, the signaling generator 2710, the information indicating the SDP offer confirmed by IP-PBX2604, flag, numeric and described how the text or the like may be added to the signaling message.
Further, in the terminal position information confirming unit 2708, and the called UE (UE 102), for example, the same corporate telephone network (here under IP-PBX2604) has been determined to be present outside the. In this case, the signaling generator 2710, the description contents of SDP offers signaling message may be a new written or overwritten to include codec mode (or codec mode set) or the like used for normal call. In this case, the signaling generator 2710, the information indicating the SDP offer confirmed by IP-PBX2604, flag, numeric and described how the text or the like may be added to the signaling message.
New or modified signaling message via the transmission unit 2702, and transmitted to the destination of the signaling message (UE 102).
The called UE which has received the signaling message (UE 102) is the own apparatus connection destination (radio access network) is determined to be a radio access network constituting a tissue LAN such as companies, and received signaling message and it identified that the SDP offer is already confirmed by the IP-PBX. In this case, the called UE (UE 102) is, SDP offer with reference to (IP-PBX2604 overwritten or newly created SDP offer in), the codec mode (or codec mode used in the call in the same within the telephone network company to select the set), and the like. At this time, in the same manner as in the third embodiment, the called UE (UE 102) may codec mode (or codec mode set) also using the presence or absence of the possibility of handover may be selected.
On the other hand, the called UE (UE 102) is the own apparatus connection destination (radio access network) is determined to be a radio access network constituting a tissue LAN of a company or the like, the SDP offer received signaling message and it was not able to identify that it is already confirmed by the IP-PBX. In this case, the called UE (UE 102) from among the candidates of the codec modes described in the SDP offer (or codec mode set), the codec mode (or codec mode set) used for ordinary communications or normal call or the like may be selected.
Further, when the connection destination of the called UE (UE 102) has an external radio access network constituting a tissue LAN such as corporations, in the called UE (UE 102), similar to the first to third embodiments codec mode (or codec mode set) selection method is applied.
Then, the called UE (UE 102) generates an SDP answer that includes a codec mode selected (or codec mode set), and transmits to the calling UE (UE 100).
Thus, in this embodiment, IP-PBX2604, when the radio access network called UE is currently connected is under the IP-PBX2604, shown in SDP offer notified from the calling UE a plurality of codec modes (or codec mode set) change candidate (new or overwritten) that generates a new SDP offer. Then, the called UE has a plurality of codec modes (or codec mode set) included in the SDP offer the changed that have changed in IP-PBX2604 from the candidates, the codec mode used in communications with the calling UE to select (or codec mode set).
Thus, when the radio access network to which the called UE is currently connected is under the IP-PBX2604 is a candidate or the like of a codec mode that is described in the SDP offer (or codec mode set), IP-PBX2604 suitable codec mode for the UE existing under (or codec mode set) (e.g., the codec mode used in the call in the same in the telephone network company (or codec mode set), etc.) may be limited to. Thus, it is possible to avoid a complex SDP negotiation between the calling UE and the called UE.
Furthermore, IP-PBX2604 transmits information indicating whether or not to confirm the SDP offer to (whether confirmed) to the called UE. Then, the called UE, when specified to be a verified SDP offer by IP-PBX2604, multiple codec modes included in the SDP offer the changed that have changed in IP-PBX2604 (or codec mode from a set) candidate, selects the codec modes to be used to communicate with the calling UE (or codec mode set).
Thus, whether the UE exists under the IP-PBX2604 (same enterprise telephone network) is accurately determined. Then, for example, if the UE is present under the IP-PBX2604 (same enterprise telephone network), UE is limited to the same codec mode (or codec mode set) used for the call in the telephone network companies it can be selected codec mode (or codec mode set). For example, when the UE does not exist under the IP-PBX2604 (same enterprise telephone network), UE is limited to the normal communication codec mode used in the (normal call) (or codec mode set), etc. it is also possible to select the codec mode (or codec mode set).
In this way, in this embodiment, UE existing in the same enterprise telephone network be accurately determined, UE can select a suitable codec mode (or codec mode set) or the like.
In the present embodiment, it has been described a case where a new description or overridden by IP-PBX2604 respect SDP offer from the calling UE. However, in the present embodiment, when UE102 is that existed in the same enterprise telephone network (here under IP-PBX2604 are), with respect to SDP answer from the called UE, novel IP-PBX2604 as described above description or may be overwritten.
Furthermore, UE 100, UE 102, and, when in IP-PBX2604 SDP offer or SDP answer is newly created or overwritten, as in the sixth embodiment, may be used the operator policy.
In this embodiment, relates to the RTP payload format shown in the fourth and fifth embodiments, the RTP payload (header, or part of the data portion) of all RTP packets for the way of giving unnecessary fields including describing .
For example, the fields of the codec mode request shown in the fourth and fifth embodiments (Codec Mode Request), or mono / stereo switching (specified number of channels) and (specified delay) Switching delay mode such as the communication partner there is a field or the like that request. These are only if the request is generated, it is harmless if the RTP payload (header, or part of the data portion).
Incidentally, described in Non-Patent Document 3 or Non-Patent Document 4 SID (Silence Insertion Descriptor), No Data, speech lost is, as a part or the like of the Frame Type Index according to such FIGS. 19 to 23, RTP payload (header may not be included in the section). In this case, these SID, No Data, only when a request to send a like speech lost occurs, the request, should I include the RTP payload (header, or part of the data portion).
An example of a RTP payload format in the case of utilizing the Frame Type Index shown in Figure 23A shown in FIG. 28. Although the Frame Type Index in Figure 23A does not include above the SID, here, for example, the SID as 8 Frame Type Index is set. Figure 28A shows the field (Codec Mode Request field, or, for mono / stereo field requesting the communication switching of the delay mode partner) an example of an RTP payload format in the case does not exist. Figure 28B shows an example of a RTP payload format when the field is present.
Here, the header portion of the RTP payload shown in FIG. 28A is a fixed length, such as 1 octet, for example. In the header portion shown in Figure 28A, "field indicating the presence or absence of a subsequent header portion" of fixed length, the field of "other" includes "Frame Type Index". Here, "header", a "next header portion" may be interpreted as part of the RTP payload data section.
Shown in Figure 28A in the "field indicating the presence or absence of a subsequent header", after the fixed-length header portion shown in FIG. 28A, information indicating whether another fixed-length header portion followed by, for example, a flag or value, etc., enter in the form of. In Figure 28A, since another header that follows is not present, the "field indicating the presence or absence of a subsequent header portion", another header that follows the information indicating that "no" is entered.
In the field "Other" shown in FIG. 28A, include other fields than both the "field indicating the presence or absence of a subsequent header portion" and a "Frame Type Index".
The "Frame Type Index" shown in Figure 28A, enters the Frame Type Index shown in Figure 23A for example.
The data portion of the RTP payload shown in FIG. 28A, for example, as shown in the fifth embodiment, is selected by the encoder side, the appropriate bit rate and combinations, the combination information and the like from entering.
On the other hand, the "field indicating the presence or absence of a subsequent header" shown in Figure 28B, information after the header portion of a fixed length shown in FIG. 28B, shows that another fixed-length header portion (next header) is "present" It enters.
It is shown in Figure 28B field and "Other", "Frame Type Index" is equivalent to Figure 28A.
Subsequent header shown in FIG. 28B is a fixed length, such as 1 octet, for example. The subsequent header shown in FIG. 28B, includes the field "type of subsequent header", "content of the Next Header", and "Other".
The "type of next header" shown in FIG. 28B, whether the next header is indicative of what is information indicating, represented by for example a flag or numerical values. For example, the "type of the subsequent header" is, Codec Mode Request, switching request number of channels such as mono / stereo, or may indicate switching request delay mode, also shows a redundant data for packet loss concealment good.
The "Contents of next header" shown in FIG. 28B, contents corresponding to the type indicated in the "type of next header" enters. For example, if the "type of next header" is Codec Mode Request is a value indicating the codec bit rate of requesting the communication partner (e.g., Frame Type Index etc. shown in FIG. 23A) enters. Further, when the "type of next header" is switching request such a channel number switching request or delay mode, such as mono / stereo switching destination information requesting (e.g., switching to "Mono" indicating information, or information, etc.) from entering indicating the switching to the "delay a". Also, if the "type of next header" is redundant data, data length and the like of redundant data entry.
In the case where the "type of the next header" shown in FIG 28B the "content of the Next Header" can be uniquely determined, may not follow the header. Although the field "type of next header" shown in FIG. 28B is a fixed length, field length of "Content of the subsequent header" may take different values ​​depending on the "type of the subsequent header".
In the field "Other" shown in FIG. 28B, although other fields other than both the "type of next header" and "contents of next header" enters, may if necessary not without.
The data portion of the RTP payload shown in FIG. 28B, for example, as shown in the fifth embodiment, is selected by the encoder side, the appropriate bit rate and combinations, the combination information and the like from entering. Further, the data unit, when the "type of the subsequent header" was redundant packet, also enters the redundant packets. In this case, encoder, as the main data to generate and select the appropriate bit rate and combinations fit into a length obtained by subtracting the length of the redundant packet from the length of the data unit bit stream.
Thus, in the first example of the present embodiment, the RTP payload consists of a header portion and a data portion, the header portion includes a header portion always provided in all RTP payload, always provided in all RTP Payload the are includes a header section not limited (next header portion shown in FIG. 28B). Further, as shown in FIGS. 28A and 28B, the always header portion provided in all RTP payload includes a field that indicates whether the subsequent header portion is provided (field indicating the presence or absence of a subsequent header) is .
Incidentally, RTP payload format in the present embodiment may take another configuration.
Figure 29 shows a second example of the RTP payload format in the present embodiment. In this embodiment, in particular all the information that needs to carry the RTP payload, assuming that carried as part of the data portion of the RTP payload, called information other than encoded data and signaling (or signaling unit). In this example also, information indicating the bit rate of the codec (Frame Type Index, etc.) is assumed that no explicitly included in the RTP payload.
Figure 29A illustrates only field that indicates the presence or absence of the subsequent signaling unit, when present as a signaling section, an example of RTP payload format. Figure 29B shows the case where the field and subsequent signaling unit indicative of the presence or absence of the subsequent signaling unit is present, an example of RTP payload format.
Here, the RTP payload shown in FIG. 29A, the fixed length from the most significant bit (e.g. 1-bit, etc.), include "field indicating the presence or absence of the subsequent signaling unit". Here, "a fixed length from the most significant bit", the "trailing signaling unit" may be part of the RTP payload data part, as described above, may be a header portion of the RTP payload .
The "field indicating the presence or absence of the subsequent signaling unit" shown in FIG. 29A, after this field, information indicating whether another signaling portion is followed by, for example, enters in the form of such flag or numerical. For example, if this field is 1 bit, if there are "0" in this field, signaling that follow are not present, if there are "1", the signaling unit for subsequent exists.
In Figure 29A, since another signaling section that follows does not exist, the "field indicating the presence or absence of the subsequent signaling portion" information separate signaling unit for subsequent indicates that "no" (i.e. in this example "0 ") to enter.
On the other hand, the "field indicating the presence or absence of the subsequent signaling unit" shown in FIG. 29B, after the signaling of the fixed length shown in FIG. 29B, another fixed length, or variable signaling portion (trailing signaling) is "present "information (in other words this example" 1 ") indicating that the entry.
Subsequent signaling unit shown in FIG. 29B is a fixed or variable length, including the field "content of the subsequent signaling" "type of subsequent signaling" and.
In "kind of subsequent signaling" shown in FIG. 29B, how the subsequent signaling is indicative of what, information indicating, represented by for example a flag or numerical values. For example, "type of subsequent signaling" is, Codec Mode Request (request for switching the codec bit rate, switching request number of channels, such as switching request bandwidth) may indicate, also, the redundant data for packet loss concealment it may indicate the presence of. Further, "the type of the subsequent signaling" is, SID, No Data, may indicate Speech lost according to such FIGS. 19 to 23.
The "Contents of the subsequent signaling" shown in FIG. 29B, contents corresponding to the type indicated in the "type of subsequent signaling" enters. For example, if the "type of subsequent signaling" is Codec Mode Request codec bit rate switching request, a value indicating the codec bit rate of requesting the communication partner (e.g., Frame Type Index etc. shown in FIG. 23A) enters. Further, when the "type of subsequent signaling" is Codec Mode Request channel number switching request is the "content of the subsequent signaling", the switching destination of the information requested (e.g., switching to "Mono" information indicating, information indicating the switch to the "stereo") enters. Also, if the "type of subsequent signaling" is the presence of redundant data, the "content of the subsequent signaling" the data length or the like of redundant data entry. In the case "contents of the subsequent signaling" is variable, depending on the type indicated in the "type of subsequent signaling", its length may vary.
In the case where the "type of the subsequent signaling" shown in FIG 29B can be uniquely determined the above "content of the subsequent signaling", or, if there is no need to determine the "content of the subsequent signaling", "contents of the subsequent signaling" is not it may be. This, for example, the "type of the subsequent signaling" is, SID, No Data or a case showing a speech lost according to such FIGS. 19 to 23. Also, while the field of "kind of subsequent signaling" shown in FIG. 29B is a fixed length, field length of "Content of the subsequent signaling" may take different values ​​depending on the "type of subsequent signaling." When the "content of the subsequent signaling" takes different values ​​depending on the "type of the subsequent signaling" is subsequent signaling itself is variable.
Moreover, RTP payload format in FIG. 29B, after the "type of subsequent signaling" and "content of the subsequent signaling", "field indicating the presence or absence of the subsequent signaling portion" may be provided. Then, RTP payload format is yet another "type of subsequent signaling", "contents of the subsequent signaling", (not shown) which may continue to "field indicating the presence or absence of the subsequent signaling unit".
The data portion of the RTP payload shown in FIG. 29B, for example, as shown in the fifth embodiment, is selected by the encoder side, the appropriate bit rate and combinations may contain such combination information. Further, the data unit, when the "type of the subsequent signaling" were redundant packet, also enters the redundant packets. In this case, the encoder selects an appropriate bit rate and combinations fit into a length obtained by subtracting the length of the redundant packet from the length of the data unit, as the main data, to generate a bit stream. In the case "type of subsequent signaling" were SID, the data portion of the RTP payload includes data generated when the SID. In contrast, "the type of the subsequent signaling" is No Data or when was the speech lost according to such FIGS. 19 to 23, the data portion of the RTP payload, the data is not included (data portion itself there may not be present).
In this example, information indicating the bit rate of the codec (such as Frame Type Index) was assumed that no explicitly included in the RTP payload. In this case, calculating the RTP payload length on the receiving side (from the data length information contained in the IP header or UDP header, calculates the RTP payload length, or the RTP payload length itself measured), and the length of the signaling unit of Figure 29 from which by removing of the codec bit rate is calculated.
Further, in the example of FIG. 29 shows an example of entering the head of the RTP payload as a fixed length from the most significant bit "field indicating the presence or absence of the subsequent signaling portion", this field is where (e.g. RTP payload of the RTP Payload Finally) it may be included in the. In this case, "field indicating the presence or absence of the subsequent signaling unit" is a means that does not mean "indicating the presence of subsequent signaling unit", "indicates whether the signaling unit to the beginning of the RTP payload".
Thus, in the second example of the present embodiment, RTP payload consists of a data portion, the signaling unit contained therein. Then, in such RTP payload, the signaling unit is always "field indicating the presence or absence of the subsequent signaling portion" provided in all RTP payload, illustrated in the signaling unit (FIG. 29B that not always provided in all RTP Payload with trailing signaling) and.
Figure 30 shows a third example of the RTP payload format in the present embodiment. In this example, a fixed length from the most significant bit of the RTP payload, by providing a field that indicates whether the header section is present, switches whether there is a header portion to the RTP payload. Accordingly, if unnecessary fields including RTP payload of all RTP packets (signaling) occurs, by using the header portion of the RTP payload includes information (signaling) to the header portion.
Figure 30A illustrates only field indicating the presence or absence of the header part, when present as a signaling section, an example of RTP payload format. Figure 30B shows the case where the header part is present, an example of RTP payload format.
Here, the fixed length from the most significant bit of the RTP payload shown in FIG. 30A (e.g. 1-bit, etc.) may include "field indicating the presence or absence of the header part". The "field indicating the presence or absence of the header portion" shown in FIG. 30A, information indicating whether the header section in the RTP payload is present, for example, enters in the form of such flag or numerical. For example, if this field is 1 bit, the header portion if there are "0" in this field does not exist, "1" is the header portion is present if placed.
In Figure 30A, since the header portion is not present, the "field indicating the presence or absence of the header portion", information (in other words the present example "0") indicating that that header is not present to enter. Incidentally, as shown in FIG. 30A, "field indicating the presence or absence of the header part" is to indicate that that the header does not exist, the "field indicating the presence or absence of the header portion", one data unit of the RTP payload and parts.
On the other hand, the "field indicating the presence or absence of header" shown in FIG. 30B, the header portion shown in FIG. 30B is "present" information (in other words this example "1") indicating that the entry. If the header is present, FIG. 30B, FIG. 30C, as shown in FIG. 30D, the header portion is composed of "extension header" and "basic header". If the header is present, the base header is always present, extension headers, the contents of the basic header information, may not exist.
Figure 30C, shows an example of the basic header. In this example, if the header part is present, "field indicating the presence or absence of header" is but an example to be included as part of base header, without this field, a part of the header portion, a data portion it may be part of. That is, the "field indicating the presence or absence of the header portion" may not be included in the basic header. The basic header has a fixed length, the remaining basic header, such as "mono / multi" field contains "Frame Type" field, and "next header whether" field.
The "mono / multi" field, for example, data carrying the RTP payload is either mono (1 channel) of the information indicating whether the multi-channel of the like stereo, for example, enters in the form of such flag or numerical. For example, if this field is 1 bit, this field "0" indicates a mono- if placed, showing a multi-channel if there are "1".
The "Frame Type" field, for example, if the "mono / multi" field described above represents a mono-, as Frame Type Index according to such FIGS. 19 to 23, information of the codec bit rate, the aforementioned SID (Silence Insertion Descriptor), it includes information such No data. In the case where "mono / multi" field described above represents a multi-channel, the "Frame Type", for example, multi-channel information, and the like bit rate. The multi-channel information, i.e., (for example 2-channel or method for encoding and right and left separately, method or, like attaching a 2 channel expansion unit) Channels or multi-channel encoding method information indicating the it is.
The "Next Header presence" field, for example, after the basic header, information on whether the extension header is present, for example, enters in the form of such flag or numerical. For example, if this field is 1 bit, indicating the field to "0" indicates that the next header is not present if put, "1" be next header exists if placed.
Base header is a fixed length, the content indicated by the "mono / multi" field, i.e., depending on whether a mono or multi-channel may have different values ​​of the fixed length.
Figure 30D, when the next header is present, shows an example of the extension header. Extension header has a fixed length, including for example "Header Type" field, the "information" field, and "next header whether" field.
The "header type" field, information indicating the type of the extension header, for example, enters in the form of such flag or numerical. For example, if this field is 3 bits, if put in this field for example, "000" indicates that the type of the extension header is "redundant data information". If placed within "header type" field "001" indicates that the type of the extension header is "Codec Mode Request (bitrate)". If placed within "header type" field "010" indicates that the type of the extension header is "Codec Mode Request (number of channels)." If placed within "header type" field "011" indicates that the type of the extension header is "Codec Mode Request (band)".
The "information" field, depending on the content of "header type", different information is entered. For example, as described above, "Header Type" be a "redundant data information", for example, the "information" field enters the Frame Type Index which represents the bit rate of the redundant data. That is to say, as in Non-Patent Document 2, the case that contains data of a plurality of frames in a single RTP packet. For example, as described above, if the "header type" is "Codec Mode Request (bitrate)", the "information" field, enter the Frame Type Index indicating the codec bit rate to request the counterpart terminal. For example, as described above, if the "header type" is "Codec Mode Request (number of channels)", the "information" field, enter a channel number or a multi-channel encoding requests the counterpart terminal. For example, as described above, if the "header type" is "Codec Mode Request (band)", the "information" field, enter information indicating the encoding bandwidth to request the counterpart terminal.
Further, for example, information fields and multi-channel showing the multi-channel, not the basic header may be included in the extension header. Conversely, the field representing the "Codec Mode Request" and the redundant data, and the information is not an extension header may be included in the basic header.
The "Next Header presence" field, for example, after the extension header, further information on whether the extension header is present, enters in the form of such flag or numerical. For example, if this field is 1 bit, if there are "0" in this field, next header indicates that no, if "1" is placed, indicates that the next header is present.
Incidentally, the extension header is a fixed length, depending on the content indicated by the "Header Type" field may have different values ​​of the fixed length.
Header type extension header, if for example, "redundant data information" or "multi-channel information", the data portion of the RTP payload shown in FIG. 30B, suits redundant data, the number of channels, the multi-channel encoding multi channel data is also included. Further, Frame Type is No Data of base header or, if was like speech lost according to such FIGS. 19 to 23, the data portion of the RTP payload may be omitted.
Further, in the example of FIG. 30 there is shown an example to enter the head of the RTP payload as a fixed length from the most significant bit "field indicating the presence or absence of header", this field is where the RTP payload (e.g. RTP Payload Finally) it may be included.
Thus, in the third example of the present embodiment, the RTP payload consists of a header portion and a data portion, if the header is not present (Fig. 30A), if the header part is present (Figure 30B) including. Further, as shown in FIGS. 30A and 30B, at all times in all RTP payload includes a field indicating whether a header portion is provided (field indicating the presence or absence of the header portion).
As described in the fifth embodiment, in order to transport block size, etc. are determined, if RTP payload length, RTP payload length is especially carry data such as normal speech, limited to a plurality of fixed length there is. In this case, as in the present embodiment, the inclusion of Codec Mode Request to RTP payload carrying data, such as normal voice, RTP payload length, may become longer than the corresponding transmission block size.
To solve this, the terminal, RTP payload data unit is not present, or, at the timing when the data unit is sent a short RTP payload as compared to the case of transmitting normal data, send a Codec Mode Request is included in these RTP payload it may be. If the data unit is not present, or if the data unit is shorter and as compared with the case of transmitting normal data, for example, a case of using a speech lost according SID, the No Data, or 19 to 23,.
Thus, according to this embodiment, the RTP payload, after giving the required fields only when necessary, thereby selecting the combination with a suitable bit rate that will fit the length of the data unit to the encoder can.
It has been described the embodiments of the present invention.
Incidentally, Frame Type Index All codec modes are available (bit rate itself, or a combination of bit rate) is not necessary to describe, it may be defined a part of the codec mode.
In the above embodiment has described the case where the correspondence between the actual codec mode (or codec mode set) and codec mode (or codec mode set) number indicating the (numerical) is defined. The number indicating the codec mode (or codec mode set) (number) has been described as described in SDP (FIGS. 8 and 11). However, the actual codec mode (or codec mode set) may also be written directly to the SDP.
Further, FIGS. 8, 11, 14 and "method 1" described in FIG. 17 represents a particular codec name. For example, "type 1" EVS, may even AMR or AMR-WB, may be a different codec. Further, in FIGS. 8 and 14, although a description has been offered only for codec scheme 1, other method (method 2, method 3 ...) are similarly described with regard, among the SDP Offer it may be included in the.
Also, different types in a certain codec, the number of channels (mono, stereo), etc.) may be described in the SDP as a codec name, in correspondence of the codec mode sets shown in FIGS. 9 and 15 it may be defined. Here, the different types in the codec, for example, in the EVS, AMR-WB interoperable mode or AMR-WB non Interoperable mode, the audio bandwidth (Narrow Band, Wide Band, Super Wide Band, Full including the Band and the like.
In the above embodiment, as the information about the characteristics of the radio access network has been described the case of using the CSG ID or base station ID. However, as the information about the characteristics of the radio access network is not limited to, for example, cell information indicating a cell (HeNB) the UE is connected, and, like the position information indicating the position of the UE and the like.
In the above embodiment, the description has been given of the case of selecting a communication start time codec modes between UE (or codec mode set) (including bit rate and processing delay) is not limited to this. For example, it can also be applied to the above embodiment a case of selecting the ability of the UE has (e.g., display size, etc.).
In the above embodiment has been described mainly using the codec for voice. However, not limited thereto, music, sound, also respect to the image, but the present invention is applicable.
Further, the present invention is not limited to the above embodiment, but can be implemented with various modifications.
Further, in the above embodiment, the case where the present invention is configured by hardware has been described as an example, the present invention can also be also be implemented in software in cooperation with hardware.
Each function block employed in the description of the aforementioned embodiments may typically be implemented as an LSI constituted by an integrated circuit. These may be implemented individually as single chips, or may be integrated into one chip including some or all of them. Here, although the LSI, depending on differences in integration, IC, system LSI, super LSI, referred to as ultra LSI.
Further, the method of circuit integration is not limited to LSI, and may be realized by a dedicated circuit or a general-purpose processor. After LSI manufacture, utilization of a programmable FPGA (Field Programmable Gate Array), may be utilized reconfigurable processor reconfigurable connections or settings of circuit cells in the LSI.
November 10, 2010 Japanese Patent Application No. 2010-252113, filed, Japanese Patent Application No. 2010-278222, filed Dec. 14, 2010, Japanese Patent Application No. 2011-84442, filed on April 6, 2011 , and the specification that is included in the Japanese Patent application No. 2011-173910, filed on August 9, 2011, the disclosure of drawings and abstract is, are all incorporated herein by reference.
The present invention is especially suitable for terminal or the like to negotiate at the start of communication by the signaling message.
100,102 UE
104,106 eNB
108,116 P-CSCF
110,114 S-CSCF
120,122,406,408,506,508,2606,2608 radio access network 124 IP core network 126 IMS network 400,402,500,502 HeNB
600 radio reception unit 602 radio transmission unit 604 the connection destination determining unit 606 information comparing unit 608 mode determining unit 610,708,2710 signaling generator 700,2700 receiver 702,2702 transmitting unit 704,2706 signaling analyzer 706,2708 terminal position information confirming unit 812 state analysis section 814 mode change unit 2600,2602,2614 AP
2610 IP network 2612 server 2704 signaling intercept portion
At the start of communication between the correspondent terminal as a communication partner of the terminal with the terminal, a codec mode selected from a plurality of codecs mode candidates notified from the counterpart terminal to a terminal to be transmitted to the counterpart terminal,
Receiving means for receiving a first information about the characteristics of the access network the counterpart terminal is currently connected,
It said first information, a determination unit configured own device compares the second information about the characteristics of the access network currently connected, determines the communication mode between the counterpart terminal and the own apparatus,
Based on the communication mode that is determined by said determining means, from the plurality of codec mode candidate selecting means for selecting a codec mode to be used for communication with the counterpart terminal,
Terminal having a.
The judgment unit, the communication mode is the same base station, a plurality of base stations, or, the first communication mode connected with a communication network including a plurality of base stations and VPN (Virtual Private Network) backbone, a second communication mode connected to different base station through the IP core network, and, determining which of respective different third communication mode of the first communication mode and said second communication mode and,
Said selecting means, wherein the communication mode is to select the highest quality codec mode when said first communication mode, the communication mode is the case of the second communication mode, when the third communication mode to select a high-quality codec mode than,
Terminal according to claim 1.
The first information is transmitted via the network node having the processing functions for call control from the counterpart terminal to the terminal,
The network node checks the validity of the first information,
It said receiving means further receives the validity of the check result in the network node,
The determining means, based on the check result, determines whether to perform a determination of the communication mode,
Terminal according to claim 3, wherein.
It said receiving means further receives the third information that indicates whether there is a possibility of handover of the counterpart terminal,
The judgment unit, the first information, the second information, the third information, and, based on the fourth information indicating the presence or absence of possibility of its own handover, determines the communication mode ,
The third information and the fourth information, each of the counterpart terminal and the terminal, whether it is stationary apparatus, whether or not it is connected to a stationary node or a communication mode whether or not it is set to a conference call mode, it shows the,
Terminal according to claim 5, wherein.
In the same apparatus or the counterpart terminal in communication, when the change in potential for handover, a changing means for changing the codec mode currently in use, further comprising,
Said first information and said second information, CSG ID (Closed Subscriber Group Identifier), or a base station ID,
Characteristics of the access network is a radio resource amount that can be allocated per terminal,
The codec mode include bit rate and processing delays,
Selected by the selecting means, codec mode used in communication with the counterpart terminal is notified using the RTP (Real-time Transport Protocol) payload,
The RTP payload is composed of a header portion and a data portion,
The length of the header portion is changed in accordance with the codec mode used for communication with the destination terminal,
Terminal according to claim 11, wherein.
The header portion includes a first header, which is always provided in all RTP payload, a second header not always provided in all RTP payload,
Wherein the first header, the second header contains a field indicating whether or not provided,
Terminal according to claim 13, wherein.
Said selection means further among the plurality of codec mode candidate, from among the codec mode candidates corresponding to at least a portion of the operator's policies to provide communications or call service to the terminal, and the counterpart terminal to select the codec mode to be used in the communication,
Wherein the plurality of codec mode candidate notified from the partner terminal includes codec mode candidates corresponding to at least a portion of the operator's policies to provide communications or call service to the counterpart terminal,
Terminal according to claim 15, wherein.
Wherein the policy the operator uses preferentially, whether to permit system codec, codec bit-rate, sampling rate, the audio band, processing delay, number of channels, bit rate and switching the number of channels in the middle session or, Variable Bit Rate (VBR) whether to allow the operation, or include whether or not to permit the redundant frame,
The RTP payload comprises a data portion, and a trailing signaling unit not always provided in all RTP payload, always provided in all RTP payload, a field indicating the presence or absence of the subsequent signaling portion,
The codec mode is notified by using the subsequent signaling unit,
The RTP payload comprises a data section, always provided in all RTP payload and the header portion is not necessarily always provided in all RTP payload, a field indicating the presence or absence of the header portion,
The codec mode is notified by using the header portion,
Third information including the plurality of codec modes candidates party terminal has generated is transmitted via the network node having the processing function for the call control to the terminal,
The network node, the terminal may access network currently connected is a subordinate of the network node, and change the plurality of codec mode candidate indicated in the third information,
It said selection means, from the plurality of codec modes candidates included in the third information after the change, selects the codec modes to be used for communication with the counterpart terminal,
The network node further transmits confirmation information indicating whether or not to confirm the third information to the terminal,
It said selection means, to indicate that a is verified the third information the confirmation information, from the plurality of codec modes candidates included in the third information after the change, and the counterpart terminal to select the codec mode to be used in the communication,
Terminal according to claim 20, wherein.
At the start of communication between the second terminal is a first terminal and the first terminal of the communication partner, the first terminal, a plurality of codec modes candidates notified from said second terminal in the codec mode selected from the communication system to be transmitted to the second terminal,
A step wherein the second terminal is for notifying the first information about the characteristics of the access network to which the second terminal is currently connected to the first terminal,
A step wherein the first terminal, for receiving the first information,
It said first terminal, said first information, said first terminal is compared with the second information about the characteristics of the access network that are currently connected, the first terminal and the second terminal and determining a communication mode between,
A step wherein the first terminal, based on the determined the communication mode from among the plurality of codec mode candidates, of selecting a codec mode used in communication with the second terminal,
Codec mode selection method comprising a.
PCT/JP2011/005986 2010-11-10 2011-10-26 Terminal and coding mode selection method WO2012063417A1 (en)
EP20110840382 EP2640052A4 (en) 2010-11-10 2011-10-26 Terminal and coding mode selection method
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US13880806 US9401975B2 (en) 2010-11-10 2011-10-26 Terminal and codec mode selection method
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3GPP: "3GPP TS 26.101 v9.0.0, "Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec frame structure"", 3GPP
3GPP: "3GPP TS 26.114 v9.3.0, "IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling and interaction"", 3GPP
3GPP: "3GPP TS 26.201 v9.0.0, "Speech codec speech processing functions; Adaptive Multi-Rate-Wideband (AMR-WB) speech codec Frame structure"", 3GPP
3GPP: "LRTF RFC 4867, "RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs"", 3GPP
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