Method, apparatus and computer program for establishing connections in a packet-switched wireless network

【0001】

[Technical field to which the invention belongs]

[Technical field]

The present invention relates to packet-based calls in wireless networks.

【0002】

[Conventional technology]

[background]

Mobile communication systems such as mobile phones and personal communication services (PCS) systems consist of multiple cells. Each cell provides a wireless communication center where the mobile unit establishes a call with another mobile unit or wireline unit connected to a public switched telephone network (PSTN). Each cell contains a radio base station, and each base station is connected to a base station controller or mobile exchange center that controls call processing between mobile units or between mobile units and PSTN units.

【0003】

Various wireless protocols exist to define communications in mobile networks. One such protocol is the Time Division Multiple Access (TDMA) protocol, such as the TIA / EIA-136 standard provided by the Telecommunications Industry Association (TIA). With TIA / EIA-136TDMA, each channel carries frames divided into 6 time slots and supports multiple (3 or 6) mobile units per channel. Other TDMA-based systems include a Global System for Mobile Communications (GSM) that uses TDMA frames divided into eight time slots (or burst periods).

【0004】

Traditional voice-oriented radio systems, such as the TIA / EIA-136 and GSM TDMA systems, utilize circuit-switched connection paths where one line is monopolized during the connection period between the mobile unit and the mobile exchange center. Such a connection is optimal for relatively continuous communications such as voice. However, data networks such as local area networks (LANs), wide area networks (WANs), and the Internet use packet-switched connections in which communication between nodes on a communication link is by data packets. Each node monopolizes the communication link only if the node needs to send and receive data packets. Packets that provide convenient and efficient access to data networks, emails, databases, and other types of data due to the rapid increase in mobile subscribers due to the widespread use of communications over data networks such as the intranet and the Internet. An exchange wireless data connection has come to be desired.

【0005】

Multiple packet-switched wireless connectivity protocols have been proposed to provide more efficient connectivity between mobile units and data networks. One such protocol is the General Packet Radio Service (GPRS) protocol, which complements existing GSM systems. Another technology built on GPRS is the Enhanced Data Rate for Global Evolution (EDGE), which offers higher data rates. The extension of GPRS by EDGE is called Extended GPRS (EGPRS). Another variant of EGPRS is the EGPRS COMPACT technology.

【0006】

Packet-switched wireless connection protocols provide efficient access to traditional data networks such as the Internet, LAN, and WAN. Such data networks are growing for voice and other forms of real-time or streaming communication (such as video, audio, and video). Various protocols have been defined to enable such real-time or streaming communication over data networks, which are often packet-switched networks. A common packet-switched network is an Internet Protocol (IP) network.

【0007】

Session Initiation Protocol (SIP), H.323, or other types of messages may be used to establish a call on an IP network. Once the call is established, other communication protocols are used to provide reliable real-time communication. Other such protocols include Real-Time Protocol (RTP) and Resource Reservation Protocol (RSVP). However, the relatively large size of common messages, such as SIP, RSVP, or H.323 messages, can result in unacceptably large delays when trying to establish a packet-based call session over a wireless network.

【0008】

[Problems to be Solved by the Invention]

As wireless networks continue to evolve, there is a need for improved methods and mechanisms that enable voice and other forms of real-time or streaming communication over packet-switched wireless networks.

【0009】

[Means for solving problems]

[Overview]

Generally, according to one embodiment, a method for providing a call in a wireless network is to communicate a step of sending an identifier to identify the call as a packet-switched call and a control signaling in the traffic channel of the wireless network. It includes steps to establish a packet-switched call.

【0010】

Some embodiments of the invention may include one or more of the following advantages: An efficient mechanism is provided to communicate the control signaling associated with packet-switched calls on the wireless network. Such efficiency allows for relatively fast call setup and cancellation. By reducing the time required for various call services such as call setup, call cancellation, and ancillary services, user convenience is improved and more efficient use of available bandwidth in wireless networks is achieved. Provided.

【0011】

Other features and advantages will become apparent from the description, drawings, and claims below.

【0012】

BEST MODE FOR CARRYING OUT THE INVENTION

[Detailed description]

In the following description, many details are provided to provide an understanding of the present invention. However, it will be appreciated by those skilled in the art that the present invention can be realized without these detailed description and many modifications and modifications from the described embodiments are possible.

【0013】

As shown in FIG. 1, the mobile communication system 10, which may be a mobile phone or personal communication service (PCS) system, includes a plurality of cells 14, each having a base station 18. Base station 18 can communicate with mobile stations 20 on radio frequency (RF) carriers 26 (eg, mobile phones, mobile computers, personal digital assistants, other types of mobile units, etc.). The group of base stations 18 is connected to the corresponding base station controller (BSC) 11. For packet-switched communications, the BSC 11 interacts with the Data Traffic Service Node 35, which is the Serving GPRS (General Packet Radio Service) Support Node (SGSN) in one embodiment. In a further embodiment, the BSC 11 may be omitted under the condition that the base station 18 is connected to the SGSN 35.

【0014】

Although cell 14 is referred to in this description, embodiments of the present invention are equally applicable to other cell segments (eg, cell sectors). As used herein, a "cell segment" generally refers to a cell, cell sector, or other division (part) of a cell.

【0015】

The SGSN35 may be connected to a Mobile Exchange Center (MSC) 12 that provides circuit-switched wireless communications. Although not shown, the MSC12 is linked to a network of cells that may be a group of cells 14 or another group of cells.

【0016】

Packet-switched data services may be provided using channels defined by the Extended GPRS (EGPRS) protocol or the EGPRS COMPACT protocol set by the European Telecommunications Standards Institute (ETSI). References to "GPRS systems" as used herein refer to either EGPRS or EGPRS COMPACT systems. Packet-switched data services may also be provided by code division multiple access (CDMA) systems such as Wideband CDMA systems (W-CDMA) or CDMA-2000 systems. Circuit-switched services in wireless networks may be by Global Systems for Mobile Communications (GSM) or TIA / EIA-136, both Time Division Multiple Access (TDMA) technologies. Alternatively, the circuit switching service may also be based on the CDMA protocol.

【0017】

SGSN35 controls the establishment, processing, and termination of packet-switched communication with mobile station 20 in cell 14 according to the EGPRS or EGPRS COMPACT protocol. Also, according to GPRS, SGSN35 communicates with gateway GPRS support node (GGSN) 36, which provides an interface to packet-switched data network 32 on core network 37. Various types of data services such as e-mail, web browsing, and file transfer are available on the packet-switched data network 32. An example of a public data network is the Internet, and an example of a private data network is a local area network (LAN) or wide area network (WAN), which may be part of various companies (eg, business, university, etc.). is there. The illustrated embodiment shows SGSN35 and GGSN36 as nodes that provide packet switching services to wireless devices such as mobile station 20, but other types of nodes according to other techniques are employed in further embodiments. You may.

【0018】

In addition to traditional data services such as email, web browsing, and file transfer available on data network 32, voice and other forms of real-time data communication (eg, audio / video streaming) are also available on data network 32. It is possible with. For such voice or other real-time data communication, terminals 40 and 42 including a network interface controller for communicating on the data network 32 may be provided. In one example, the terminal 40 is a data networkable telephone such as the i2004 telephone by Nortel Networks Ltd. Such a data networkable telephone translates voice and other types of real-time data into packets that are communicated over the data network 32. An example of a terminal 42 is a computer that conforms to voice processing capabilities and a soft phone routine, such as Nortel's i2050 product. Therefore, users can communicate with each other on the data network 32 without going through a conventional circuit-switched network such as PSTN16.

【0019】

A terminal on the packet-switched data network 32 can also communicate with a telephone device connected to the PSTN 16 via a media gateway 33 connected between the public switched telephone network (PSTN) 16 and the packet-switched data network 32. The media gateway 33 converts between the packet-switched voice frame and the circuit-switched voice frame.

【0020】

The Media Gateway Control Function (MGCF) module 39 performs conversion between packet switching and circuit switching signaling in a call session between terminals on both sides of the media gateway 33. For example, MGCF module 39 generates Signaling System No. 7 (SS7) signaling to PSTN16. In addition, the communication system includes a Call State Control Function (CSCF) module 41 that provides full call control for packet-based communication sessions. In some embodiments, the CSCF module 41 is a SIP proxy or server that receives a call request on behalf of another entity, analyzes the logical address or identifier of the call request, and forwards the call request to its intended destination.

【0021】

An example of a data network 32 is a connectionless packet-switched network such as an Internet Protocol (IP) network. IP is described in RFC (Request for Comments) 791 in September 1981, entitled "Internet Protocol". Other versions of IP, such as IPv6, and other packet-switched standards may also be utilized in further embodiments. The version of IPv6 is described in RFC 2460 in December 1998, entitled "Internet Protocol, Version 6 (IPv6) Specification".

【0022】

Packet-switched networks, such as IP networks, communicate with packets, datagrams, or other data units on the network. Unlike circuit-switched networks that provide a dedicated end-to-end channel portion (eg, a time slot) for the duration of a call session, packet-switched networks are based on a connectionless internetwork layer. Packets or other data units injected into a packet-switched network may travel independently on any network (and possibly on a different network) to a destination point. Packets may arrive in pieces. Packet routing is based on one or more addresses carried in each packet.

【0023】

Another type of packet network is a connected packet network such as Asynchronous Transfer Mode (ATM) or Frame Relay Network. In a connected packet network, a virtual circuit or connection is established between two endpoints, and packets are delivered in the same order as they were transmitted.

【0024】

Control messages according to the Session Initiation Protocol (SIP) may be used to establish a call session on a packet-switched network such as an IP network. SIP is part of the Internet Engineering Task Force (IETF) multimedia data and control architecture. The SIP version was published in August 1999 in "SIP: Session Initiation Protocol". It is entitled "Protocol)" and is described in RFC2543. SIP is used to invite members to sessions that may have been advertised by some other mechanism, such as email, newsgroups, web pages, and other mechanisms, as well as initiate a calling session. You may. Other protocols in the IETF's multimedia and control architecture are as described in RFC 2205, Resource Reservation Protocol (RSVP) for Reserving Network Resources, RFC 1889. To control the delivery of real-time transport protocol (RTP) to transfer real-time data and provide quality of service (QoS) feedback, and streaming media, as described in RFC 2326. Real-Time Streaming Protocol Protocol (RTSP), Session Description Protocol (SDP) for describing multimedia sessions, as described in RFC2327, and multimedia sessions by multicast, as described in RFC2974. Includes Session Announcement Protocol (SAP) for advertising.

【0025】

Other standards may be adopted in further embodiments for establishing packet-switched calls on the data network 32. Such other standards may be any standard that provides interactive and real-time voice or other streaming communication on the data network 32. Another standard is the H.323 Recommendation by the International Telecommunication Union (ITU). In addition, in further embodiments, protocols may be employed that define control signaling for call sessions on ATMs or other connected data networks 32.

【0026】

A "call session" as used herein generally refers to any of the voice, video, or other real-time and interactive sessions established between two or more network elements connected to the data network 32. pointing. The network element may include terminals linked to the data network 32, such as terminals 40 or 42. Another terminal for call sessions is the phone connected to PSTN16. Another terminal for a call session on data network 32 is one of mobile stations 20 communicating over the wireless infrastructure provided by base station 18 and SGSN35. As used herein, "packet-switched call" or "packet-switched call session" refers to a call session established on a packet-switched data network 32 for exchanging real-time data such as voice and video. More generally, a "packet-based call (packet-based call)" or "packet-based call session" is established on any type of packet-based data network 32 (either connectionless or connected network). Mentions the call session to be made. In a clear description, reference is made to packet-switched calls. However, in a further embodiment, any type of packet-based call may be made.

【0027】

One technique for the mobile station 20 to establish a packet-switched call with another terminal (eg, 40,42) on the data network 32 is used on the data network 32 and the terminals 40,42 and the media. Sending call control signaling according to the call control protocol enabled by gateway 33. One such call control protocol is SIP. Other packet-switched call control protocols, such as H.323, may be employed in further embodiments. Call control signaling is communicated over a radio access network (including base station 18, base station controller 11, and SGSN35). The base station 18 and the base station controller 11 are collectively referred to as a wireless network controller (RNC). Radio access network is EGPRS or EGPRS It may be a GSM / EDGE radio access network (GERAN) operating according to the COMPACT protocol. Alternatively, the radio access network may be a UMTS (Universal Mobile Telecommunication System) terrestrial radio access network (UTRAN) according to the W-CDMA protocol.

【0028】

Call control signaling communicated over the radio access network is forwarded by the GGSN36 to the CSCF module 41. Call control signaling is received by CSCF module 41 and the call session is established after the call setup procedure has been performed. If the destination terminal is attached to PSTN16, MGCF module 39 is also included in the call setup procedure.

【0029】

In a typical SIP call setup sequence, SIP messages are text or ASCII and are carried in IP packets that include IP headers and UDP (User Datagram Protocol) headers, so SIP call control messages. Message size is very large. UDP, entitled "User Datagram Protocol" in August 1980, was described in RFC768, a transport layer for managing connections between network elements on an IP network. provide.

【0030】

Due to the relatively large message size of SIP messages for call setup, call cancellation, and ancillary services, such messages can be communicated over a radio access network, call setup, call release, and ancillary services. An efficient mechanism for performing services is provided according to some embodiments of the present invention. In one embodiment, the SIP call setup time on the wireless network generally corresponds to the call setup time performed on the GSM circuit-switched wireless network.

【0031】

In one embodiment, new MS (base station) codes are defined for Random Access Channel (RACH), Packet Random Access Channel (PRACH), and COMPACT Packet Random Access Channel (CPRACH). RACH, PRACH, or CPRACH is used by base stations to request access to radio access networks. Various RACH, PRACH, and CPRACH channels are adopted depending on the type of radio access network. RACH is used in the GSM radio access network, PRACH is used in the EGPRS radio access network, and CPRACH is used in the EGPRS COMPACT radio access network. The MS code can be 8-bit or 11-bit wide, depending on which of RACH, PRACH, and CPRACH is used.

【0032】

In one embodiment, the 8-bit MS code carried in the RACH burst is code AF.₈, AF₇, AF₆, AF₅, AF₄, AF₃, AF₂, And AF₁The code AF [8: 1] for making AMR (compatible multi-rate) full-rate packet-switched voice calls and the code AH [8: 1] for making AMR half-rate packet-switched voice calls. It contains code BF [8: 1] for answering full-rate packet-switched paging voice calls and code BH [8: 1] for answering AMR half-rate packet-switched paging voice calls.

【0033】

Similar 8-bit MS codes are defined for PRACH, code CF [8: 1] for making AMR full-rate packet-switched voice calls and code CH [8: 1] for making AMR half-rate packet-switched voice calls. 8: 1], code DF [8: 1] for answering AMR full-rate packet-switched paging voice calls, and code DH [8: 1] for answering AMR half-rate packet-switched paging voice calls. I'm out. Similarly, the 11-bit MS code is defined for PRACH and CPRACH. Two different training sequences TS1 and TS2 are used for different modulation schemes on the Um link between mobile station 20 and base station 18. The defined 11-bit MS code is TS1 EF [8: 1] or TS2 GF [8: 1] for making AMR full-rate packet-switched voice calls, or TS1 EH for making AMR half-rate packet-switched voice calls. [8: 1] or TS2 GH [8: 1], AMR Full rate packet switching TS1 FF [8: 1] or TS2 for answering paging voice calls HF [8: 1], and TS1 FH [8: 1] or TS2 HH [8: 1] for answering AMR half-rate packet-switched paging voice calls.

【0034】

The above code is provided for illustrative purposes only, as different codes may be defined to provide different services in different embodiments.

【0035】

The mobile station 20 communicates a RACH message and executes a channel request to initiate a call or answer a page. Channel requests may also be performed by PRACH or CPRACH bursts, depending on the type of system. As used herein, "random access channel" generally refers to any one of RACH, PRACH, or CPRACH. If the desired call is a packet-switched call, the RACH message contains one of the codes defined above. In response to a RACH message containing a packet-switched call code, the wireless network controller or SGSN35 allocates a predetermined logical channel combination and physical channel and executes a packet-switched call. The logical channel combination is described below in connection with FIG. 2, and the physical channel refers to the frame time slot and the carrier frequency assigned to the mobile station.

【0036】

According to some embodiments, SIP messages (and other call control messages) are EGPRS or EGPRS. Carried in a dedicated packet data traffic channel (PDTCH) burst, as defined by COMPACT. An example PDTCH burst 150 is shown in FIG. By transporting SIP messages using PDTCH mapped to a dedicated physical channel, the time for packet-switched call setup sequences is equivalent to the time for GSM circuit-switched call setup sequences. A further advantage of using PDTCH mapped to a dedicated physical channel to carry SIP messages is that the messages are transparent to a radio access network such as GERAN. That is, SIP messages carried in a dedicated PDTCH burst are treated as traffic and not processed by the radio access network. A physical channel is "dedicated" if it is assigned for use by a given mobile station and is not shared with another mobile station while assigned to a given mobile station. ..

【0037】

With reference to FIG. 2, a new logical channel combination in GERAN is defined according to one embodiment. Logical channel combinations include: TCH + FACCH + SACCH + PDTCH + PACCH + PTCCH [0038]

A TCH is a logical channel (mapped to the same dedicated physical channel as the PDTCH) used to carry voice bearer data to an optimized voice bearer. FACCH is the first related control channel. SACCH is a slow-related control channel. PACCH is a packet-related control channel. PTCCH is a packet timing control channel. As mentioned above, SIP message 100 associated with other control messages such as RSVP message 102, RTCP message 104, and DTMF (dual tone multi-frequency) message 106 is carried in a dedicated PDTCH burst 108. In other embodiments, instead of using a dedicated physical channel, the PDTCH is transported over a shared physical channel (shared time slot or frame time slot) due to increased post-dial delay time. Will be done.

【0039】

In one arrangement, referred to as an optimized voice bearer arrangement, the voice frame 110 is carried in a dedicated traffic channel (TCH) burst 112. In addition, the SID (silence descriptor) frame 114 transmitted over the wireless network during the dead period is also carried in the TCH burst 112. In another arrangement, instead of transporting the audio frame 110 and SID frame 114 of the TCH burst 112, the audio frame 110 and SID frame 114 are conveyed in the PDTCH burst 108. This other arrangement is referred to as a shared arrangement.

【0040】

Radio resource (RR) management message 116 is carried in PACCH burst 118, SACCH burst 120, or FACCH burst 122. PACCH burst 118 is used to carry RR message 116 during call setup. SACCH Burst 120 and FACCH Burst 122 carry relevant post-call setup control messages during a call session. The timing forward / backward message 124 is carried in the PTCCH burst 126. Such messages are not transparent to the radio access network. In another embodiment, some of the RR messages 116 may also be delivered in a dedicated PDTCH burst 108.

【0041】

If the radio access network is UTRAN, another logical combination (called a forwarding channel combination) is defined to map the above messages and signals to a UTRAN-specific physical channel. Packet-switched call control signals such as SIP, RSVP can be assigned different channelized codes and / or scrambled sequences other than those assigned to voice bearer data.

【0042】

In one embodiment, two groups of radio bearers (RBs) are defined: a first group (eg, RB5 ~ 31) and a second group (eg, RB0 ~ 4, or known as signaling RB or SRB). Will be done. The first group of RBs are used to communicate the data attached to the Primary Packet Data Protocol (PDP) context. The PDP context may contain the following information: The PDP type can be identified to identify IP, X.25, or PPP (Point-to-Point Protocol) as the packet data protocol. The PDP address is also included in the PDP context because it is a quality of service (QoS) profile that identifies the QoS profile requested or navigated for a given flow. When the communication session is first established, the primary PDP context containing the default QoS profile is launched.

【0043】

The first group of RBs are used to carry the data attached to the QoS identified in the primary PDP context, and the second group of RBs (SRBs) are used to signal by already reserved QoS requirements. Transport data. Therefore, in the example of FIG. 2, the PDTCH burst 108 and the TCH burst 112 are carried in the RB of the first group, and the control signaling (RR message 116 and timing forward / backward message 124) is carried out in the RB of the second group ( Transported at SRB).

【0044】

One of the following logical channel combinations is assigned depending on whether a full-rate call or a half-rate call is set up. TCH / F + FACCH / F + SACCH / F + PDTCH / F + PACCH / F + PTCCH (full rate call), and TCH / H + FACCH / H + SACCH / H + PDTCH / H + PACCH / H + PTCCH (half) Rate call) [0045]

The PTCCH burst only needs to be transmitted or configured every 16 26 multiframes. Multiframes are used to communicate control and traffic signaling between mobile stations and base stations. In one arrangement, each multi-frame contains 52 frames, and each frame contains 8 time slots. Multi-frame starts at frame FN0 and ends at frame FN51. One block is composed of four frames. In some embodiments, PTCCH may not be required. Also, in some embodiments, both FACCH and PACCH may not be required (one or the other is sufficient).

【0046】

An efficient mechanism by using a PDTCH mapped to a dedicated physical channel to carry packet-based control signaling and a TCH mapped to the same physical channel to carry voice bearer data. Is provided to set up or cancel a packet-switched call session on a radio access network such as GERAN or UTRAN.

【0047】

Referring to FIG. 3, a simplified message flow for setting up a packet-switched call between a mobile station 20 and a terminal (eg, 40, 42, or 33) on a packet-switched data network 32 is illustrated. There is. Figures 7A-7C describe a more detailed call setup flow (see below). Figure 3 shows the exchange of messages between a mobile station and a wireless network controller (RNC). The RNC forwards the control message to the SGSN35. Control messages regarding the radio access network are processed by the SGSN35, and SIP messages are communicated to the data network 32 and processed by the SIP proxy and one or more terminals associated with the call.

【0048】

According to one embodiment, to initiate a call, mobile station 20 sends a RACH burst (at 202) to the RNC to perform a channel request. Mobile station 20 also sends a RACH burst to respond to the page. For packet-switched calls, RACH contains one of the packet-switched MS codes above. Instead of RACH, PRACH or CPRACH bursts may also be used. In contrast, the RNC communicates (in 204) a dedicated physical traffic channel or time slot that can support logical channel combinations and physical channel allocation for packet-switched calls over an authorized channel. To do. The allocation of logical channel combinations allows the mobile station 20 to carry out SIP message communication using PDTCH mapped to a dedicated physical channel.

【0049】

The mobile station then (in 206) sends an initial PDTCH burst containing a SIP Invite request, such as included in the IP packet by the associated IP and UDP headers. The Invite request contains the destination address of the terminal being called and indicates that the called terminal is invited to join the call session. The RNC sends the burst to the SGSN35, which sends the SIP Invite message contained in the IP packet to the CSCF module 41 via the GGSN36. The Invite request is processed by the CSCF module 41, which analyzes the logical address included in the Invite request and identifies the location of the destination terminal. Depending on the location of the destination terminal, the CSCF module 41 generates signaling and establishes a call on the data network 32 or PSTN16.

【0050】

The SIP Ringing response is returned from the called party once it has been allocated and received an Invite request. The Ringing response is returned via the CSCF modules 41, GGSN36, SGSN35, and RNC, which transmit the Ringing response in another PDTCH burst (at 208) to mobile station 20. When the call is received by the destination terminal, the destination terminal returns a SIP OK response, which is forwarded by the base station (at 210) in a PDTCH burst. In response to the OK response, the mobile station sends a PDTCH burst carrying the SIP ACK message (at 212) and recognizes the OK response. At this point, the call is set up and voice or other real-time data is being communicated. In addition, the RR management signaling carried in the PACCH is exchanged between the mobile station and the RNC (at 214) to perform adjacent cell signal strength measurements.

【0051】

In addition, for optimized voice bearers, certain parameters used to build RTP / UDP / IP headers by both mobile stations and RNCs (eg UDP port numbers, IP destination addresses, etc.) are PACCH burst uplinks and Sent to both downlinks.

【0052】

Referring to FIG. 4, when a call session, such as an RTP call session, is set up (at 302) using the mechanism described above, one of the associated terminals can cancel the call. In the example of FIG. 4, the mobile station can start canceling the call. To do so, the mobile station sends a PDTCH burst carrying SIP Bye messages (at 304) to the RNC, which forwards the PDTCH burst to the SGSN35. The Bye request is processed by the CSCF module 41, which forwards the Bye request to the destination terminal. The target terminal is SIP Send the OK response back to SGSN35 via CSCF module 41 and GGSN36. The OK response is communicated to the RNC, which communicates the OK response in the PDTCH burst to the mobile station (at 306). Following the receipt of the OK message from the RNC, the mobile station sends a PACCH burst with appropriate RR management signaling (at 308) to indicate to the radio access network that the call session has ended. This allows the radio access network to release the channels used during the call session for other calls.

【0053】

In addition to setting up and canceling calls, PDTCH bursts are also used to carry SIP signaling to perform ancillary services such as call hold, conferencing with multiple parties. With reference to FIG. 5, an example of a call hold sequence is illustrated. An RTP call session (at 402) is established between the mobile station and the remote station over the radio access network. The remote terminal puts the mobile station on hold, which is achieved by sending a SIP Invite request with c = 0 (ie 0.0.0.0 in some exemplary arrangement with an IP address). This is carried from the RNC to the mobile station in the PDTCH burst. The mobile station recognizes that an Invite request with an OK message is sent back to the base station in a PDTCH burst at 406 and also forwarded to the remote terminal via the radio access network and data network 32. Then the remote terminal is SIP It returns an ACK message, which is communicated by the RNC to the mobile station (at 408) in the PDTCH message carrying the ACK message. At this point, the RTP call session is interrupted (at 410).

【0054】

Once the remote terminal is ready to put the mobile station on hold, it sends another Invite request (including the actual IP address), which is forwarded to the mobile station by the base station in a PDTCH burst (in 412). To. The mobile station returns an OK request carried in a PDTCH burst (at 414). Upon receiving an OK response, the remote terminal sends an ACK request, which is communicated by the base station in a PDTCH burst (at 416). At this point, the RTP call session is reestablished (at 418).

【0055】

With reference to FIGS. 7A-7C, a more detailed message flow for performing a call setup between the mobile station and the endpoint connected to the data network 32 is illustrated. The mobile station first performs a radio resource control (RRC) connection setup (in 602) and an Iu signaling connection setup by RNC (in 604). The mobile station then performs the GPRS attach procedure (at 606). An attach procedure is performed to inform the radio access network that the mobile station is available. To activate the primary PDP context, the mobile station sends an Activate PDP Context request (at 608) to the RNC, which forwards the request (at 610) to the SGSN35. In response, the SGSN35 performs a radio access bearer allocation procedure (at 612) to assign one or more radio access bearers to the mobile station.

【0056】

Once one or more radio bearers are set up, SGSN35 sends a Create PDP Context request (at 614) to GGSN36. GGSN36 responds to SGSN35 (at 616) with a Create PDP Context request. When the SGSN35 receives the Create PDP Context response, the SGSN35 sends an Activate PDP Context Accept message to the RNC (at 618), which forwards the Accept message to the mobile station (at 620). At this point, the primary PDP context has been activated and a radio bearer has been assigned. To this point, the procedures performed on 602-620 follow the general setup procedures performed on mobile stations, RNC, SGSN35, and GGSN36.

【0057】

Once the primary PDP context is established, PDTCH bursts can be used to carry various types of control signaling, including SIP signaling and other signaling. For example, as described below, a mobile station uses Dynamic Host Configuration Protocol (DHCP), which contains configuration information from a DHCP (Dynamic Host Configuration Protocol) server. Can be done (at 624). The configuration information includes the IP address of the mobile station. DHCP was described in RFC1541 in October 1993, entitled "Dynamic Host Configuration Protocol". DHCP messages are carried in PDTCH bursts between the mobile station and the RNC, according to some embodiments.

【0058】

During DHCP procedure 624, the mobile station receives the domain name of CSCF module 41. To find the IP address of CSCF module 41, the mobile station sends a DNS-Query (at 626) to the DNS server. In response, the DNS server sends back (at 628) a DNS-Response containing the IP address of CSCF module 41 to the mobile station. DNS-Query and DNS-Response messages are carried in PDTCH bursts according to some embodiments.

【0059】

SIP Register requests are commonly communicated to allow call establishment using SIP. The SIP Register request may be sent, for example, to the known "all SIP servers" multicast address "sip.mcast.net". However, because the address of CSCF module 41 is known, the mobile station sends a SIP Register request (at 630) to CSCF module 41. A Register request contains a "To" field that contains the address of record for which the registration is created or updated, a "From" field that contains the address of the entity responsible for the registration, and the registration request. It may contain multiple fields, including a "Request URI" field that means a destination. In response to the SIP Register request, CSCF module 41 registers the location of the mobile station and sends a SIP OK response back to the mobile station (at 632).

【0060】

At this point, the mobile station can perform a call setup initiated by the mobile station that sends a SIP Invite request (at 634) to CSCF module 41. In some embodiments, the Invite request is communicated between the base station and the RNC in a dedicated PDTCH burst. The CSCF module 41 forwards the SIP Invite request (in 636) to the media gateway 33 (Figure 1) or a network endpoint attached to the data network 32, which may be another endpoint. In response, the network endpoint (referred to as the media gateway in this example) sends a SIP 183 Session Progress message (in 638) and the network endpoint has sufficient resources to answer the call request. Show that you are doing. SIP 183 Session Progress messages are forwarded to the mobile station by CSCF module 41 (at 640). SIP 183 Session between RNC and mobile station Progress messages are carried in a dedicated PDTCH burst.

【0061】

In response to the SIP 183 message, the mobile station sends a provisional permit (PRACK) response back to CSCF module 41 (in 642). The PRACK response is forwarded to the network endpoint (at 644). The network endpoint then sends a 200OK response back to the CSCF module 41 (at 646), which forwards the 200OK response (at 648) to the mobile station. PRACK and OK messages are also carried in a dedicated PDTCH burst between the RNC and the mobile station.

【0062】

To establish the downlink flow with the desired quality of service (QoS), the network endpoint sends an RSVP PATH message (at 650) to the GGSN36. The RSVP PATH message contains Sender_Tspec information that identifies the traffic profile expected to be generated by the network endpoint. The GGSN36 forwards the RSVP PATH message (at 652) to the mobile station. In some embodiments, RSVP messages are carried between the RNC and the mobile station in a dedicated PDTCH burst.

【0063】

If the primary PDP context already established for the mobile station cannot provide the QoS required in the RSVP PATH message, the mobile station can establish a secondary PDP context. The primary PDP context allows the mobile station to activate one or more secondary PDP contexts with different QoS profiles, if desired. The secondary PDP context invokes the procedure, and the radio access bearer assignment and radio bearer setup procedure is performed between the mobile station and the GGSN36 (at 654).

【0064】

Once the secondary PDP context providing the desired QoS level is established, the mobile station (at 656) sends an RSVP RESV message in response to the RSVP PATH message. The RSVP RESV message is received by the GGSN36, which forwards the RSVP RESV message (at 658) to the network endpoint. Again, RSVP RESV messages are carried in a dedicated PDTCH burst.

【0065】

The RSVP process described above is also performed in the reverse direction to establish the uplink with the desired QoS. This is illustrated by the RSVP PATH message sent and transmitted by the mobile station (at 659). In response, the network endpoint sends an RSVP RESV message (at 660). The procedure for changing the secondary PDP context based on the downlink RSVP process may also be performed if necessary.

【0066】

The mobile station then sends a COMET message (at 661) to CSCF module 41 to indicate that the condition has been met. COMET messages are forwarded to the network endpoint (at 662). In response, the network endpoint sends a 200OK message (at 664) to CSCF module 41, which forwards a 200OK response (at 666) to the mobile station. COMET and 200OK messages are carried between the RNC and the mobile station in a dedicated PDTCH burst.

【0067】

At this point, the network endpoint is ready to respond to the mobile station. The SIP 180 Ringing response is sent to CSCF module 41 (at 668), which forwards the Ringing response (at 670) to the mobile station. The PRACK message is then sent from the mobile station (at 672) to the CSCF module 41, which forwards the PRACK message (at 674) to the network endpoint. In response, the network endpoint returns a 200 OK (PRACK) response (at 676), which is forwarded to the mobile station by the CSCF module 41 (at 678). Following this, the network endpoint also sends (at 680) 200 OK in response to the SIP Invite request communicated on 634. The 200OK response is forwarded to the mobile station by CSCF module 41 (at 682). The mobile station then SIPs (at 680) The Ack message is sent back to CSCF module 41, which forwards the Ack message (at 682) to the network endpoint. Messages exchanged between 660 and 682 are carried in a PDTCH burst.

【0068】

When an Ack response is received, an RTP bearer path is set up between the mobile station and the network endpoint (at 684). Voice and other real-time traffic is carried between mobile stations and network endpoints on the RTP bearer path.

【0069】

The following describes the performance of packet-switched calls using techniques according to some embodiments that correspond to GSM circuit-switched calls. The traffic flow through the cell site is defined as the product of the number of calls N during a particular time period and the average duration T of the calls. In traffic theory, a commonly considered unit of time is a period of one hour. Therefore, the number of calls N is expressed with respect to the incoming call rate λ (the number of calls per unit time), and the average period T of calls is expressed with respect to the unit time per call. The traffic strength A (according to Erlang) is given by: A = λ T / 3600 where λ is the unit of calls per busy hour and T is the unit of seconds.

【0070】

The following number of channels per sector is available for the 4/12 frequency reuse pattern and the bandwidth of 15MHz. CH = (15000kHz) (8) / (12) (200kHz) = 50 [0071]

For a GSM circuit-switched call setup sequence, assume that logical channel combination VII (which requires one time slot) is used to support the mapping of SDCCH to the basic physical channel. VII SDCCH / 8 (0 ... 7) + SACCH / 8 (0 ... 7) [0072]

In addition to this, a logical channel combination IV (which also requires one time slot) is used for the beacon carrier. IV FCCH + SCH + BCCH + CCCH [0073]

Therefore, for the GSM circuit-switched call setup sequence, the number of channels per sector is 48 for the 4/12 frequency reuse pattern and the bandwidth of 15 MHz.

【0074】

For SIP call setup sequences, assume that Logical Channel Combination IV (requiring one time slot) or XI (requiring one time slot) is used for either the Beacon carrier or the COMPACT CP BCCH carrier. .. IV FCCH + SCH + BCCH + CCCHXI PFCCH + PSCH + CPBCCH + CPCCCH + PDTCH + PACCH + PTCCH [0075]

Therefore, for the SIP call setup sequence, the number of channels per sector is 49 for the 4/12 frequency reuse pattern and the bandwidth of 15MHz.

【0076】

In some examples, a system that implements a packet-switched call setup procedure using SIP according to some embodiments uses RIL3-CC, RIL3-MM, RIL3-RR, and DTAP as described below. It is more spectrally efficient than a system that implements GSM circuit-switched call setup procedures. Blocking is the inability to make a call due to the availability of a small number of channels. For example, a blocking value of 0.02 means that 2 calls are blocked for every 100 attempts made. Table 1 shows a comparison of blocking values for a typical GSM circuit-switched call setup sequence and a typical SIP call setup sequence in an exemplary arrangement.

【0077】

[table 1]<img file="JP2004505521A_D0001.tif" /> 【0078】

Therefore, as shown in the example in Table 1, systems that implement packet-switched call setup procedures using SIP have RIL3-CC, RIL3-MM, RIL3-RR, for average call durations longer than 120 seconds. And can be slightly more spectrally efficient than systems that use DTAP to implement GSM circuit-switched call setup procedures. However, the benefits of SIP call setups are diminished when considering traditional wireless resource management messages for SIP call setups. There is also some extra radio resource management signaling for invoking the secondary PDP context.

【0079】

The spectral efficiency of a typical SIP call setup / cancellation sequence is further improved by compressing the UDP / IP header into a SIP request method and a response code communicated during the general SIP call setup / cancellation sequence. Will be done. Spectral efficiency is also improved by using the coding schemes MCS-2 to MCS-9. Finally, when simple token technology is realized, the size of SIP request method and response code will be reduced by about 12%.

【0080】

With reference to FIG. 8, the components of the wireless network controller 700, the data traffic service node 35, and the mobile unit 20 are illustrated. Such components are for illustrative purposes only and do not limit the scope of the invention. In further embodiments, other architectures of such components are possible. For example, the wireless network controller 700 may actually include multiple platforms such as base stations and base station controllers. In the wireless network controller 700, the transceiver 727 is connected to the antenna tower 754 that transmits and receives the carrier 26. Transceiver 727 is connected to control unit 750 (or multiple control units) capable of performing various software routines 749. The storage unit 747 (or a plurality of storage units) may also be connected to the control unit 750. In addition, the wireless network controller 700 includes an MSC interface 752 that is linked to a link 764 (eg, a T1 link) that is linked to the MSC. The wireless access controller 700 also includes an interface 751 (in one embodiment, an Iu-ps interface according to EGPRS or EGPRS COMPACT) for communicating with the SGSN35 over a link (eg, Iu-ps link). ..

【0081】

In one embodiment, the SGSN35 includes interface units 777 and 779 for communicating over Iu-ps and Gs links, respectively. The processing core of the data traffic service node 35 includes a control unit 769 (or multiple control units). The storage unit 771 (or a plurality of storage units) is connected to the control unit 769. The routines and modules that form the data traffic system controller 740 may first be stored in storage unit 771 and loaded by control unit 769 for execution. The SGSN35 further includes an interface 781 (eg, a Gn interface) for communicating with the GGSN36 (FIG. 1). In another embodiment, the interface 781 may be a network interface controller or transceiver capable of communicating over the data network 32.

【0082】

The carrier is communicated between the antenna 754 connected to the wireless network controller 700 and the antenna 762 of the mobile station 20. In an exemplary arrangement of mobile station 20, wireless transceiver 764 is connected to antenna 762 to transmit and receive carrier 26. The control unit 766 (or multiple control units) may be coupled to one or more radio transceivers 764. The control unit 766 is connected to the storage unit 768 (or a plurality of storage units). Software routines 768 that can be run on control unit 766 may first be stored in the non-volatile portion of storage unit 768. The input / output (I / O) controller 774 is connected to the keyboard 770 and the display 772 of the mobile station 20.

【0083】

To support the communication of IP packets and SIP messages, the mobile station 20 further includes a SIP stack 702 and a UDP / IP stack 704. On the sender side, the UDP / IP stack 704 appends the appropriate UDP and IP headers for encapsulation within the IP packet. On the receiving side, the UDP / IP stack 704 extracts payload information, such as a SIP message, from the received IP packet. SIP stack 702 is a state machine that provides analysis, processing, and generation of SIP requests and responses. Other modules, including RSVP agents and modules that can generate and receive DHCP and DNS messages, are also present within mobile station 20.

【0084】

The various software layers, routines, or modules described herein may be executable on various processing elements such as the control units described above. Each control unit may include a microprocessor, a microcontroller, a processor card (including one or more microprocessors or microcontrollers), or other control or computing device. A "controller" as used herein refers to either hardware or software, or a combination of the two. "Controller" also refers to a single component or multiple components (either hardware or software and ware).

【0085】

The storage unit contains one or more machine-readable storage media for storing data and instructions. Storage media are semiconductors such as dynamic or static random access memory (DRAM or SRAM), erasable programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM), and flash memory. Various memories including memory devices, magnetic disks such as fixed disks, floppy disks, and removable disks, other magnetic media including tapes, and optical media such as compact disks (CDs or digital video disks (DVDs)). Includes morphology. Instructions that form different software layers, routines, or modules in different network elements are stored in their respective storage units. Instructions, when executed by each control unit, cause the corresponding station or system to perform a programmed operation.

【0086】

Software layer, routine, or module instructions are transferred to a station or system by one of many different methods. For example, a code segment containing instructions stored on a floppy disk, CD, or DVD medium or transferred by a network interface card, modem, or other interface device is read into the system and the corresponding software layer, routine, or module. Is executed as. In a read or transfer process, a data signal embodied in a carrier wave (transmitted over a telephone line, network line, wireless link, cable, etc.) communicates a code segment containing an instruction to a network element. Such carriers are in the form of electrical, optical, acoustic, electromagnetic, or other types of signals.

【0087】

Although the present invention has been described for a limited number of embodiments, one of ordinary skill in the art will recognize numerous modifications and changes. The appended claims are intended to cover all such amendments and modifications within the scope and gist of the present invention.

[Simple explanation of drawings]

[Figure 1]

It is a block diagram of the embodiment of the communication system according to one embodiment.

[Figure 2]

It is a figure which shows the allocation of the logical channel combination which carries a control message according to one Embodiment.

[Fig. 3]

It is a figure which shows the message flow between a mobile station and a wireless network controller (RNC) for performing a call setup according to one embodiment.

[Fig. 4]

It is a figure which shows the message flow of the sequence for canceling a call session according to one Embodiment.

[Fig. 5]

It is a figure which shows the message flow of the sequence for putting a call in a hold state according to one embodiment.

[Fig. 6]

It is a figure which shows the packet data traffic channel (PDTCH) for carrying the control signaling of a packet switching call according to one Embodiment.

[Fig. 7A to Fig. 7C]

It is a figure which shows the message flow of the call setup sequence according to another embodiment.

[Fig. 8]

It is a block diagram of the component of the communication system of FIG.