WO2009092437A1 - Réalisation d'un appel de données de télécopie et analogiques - Google Patents

Réalisation d'un appel de données de télécopie et analogiques Download PDF

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Publication number
WO2009092437A1
WO2009092437A1 PCT/EP2008/050656 EP2008050656W WO2009092437A1 WO 2009092437 A1 WO2009092437 A1 WO 2009092437A1 EP 2008050656 W EP2008050656 W EP 2008050656W WO 2009092437 A1 WO2009092437 A1 WO 2009092437A1
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WO
WIPO (PCT)
Prior art keywords
session
media type
user
towards
call
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/EP2008/050656
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English (en)
Inventor
Juha Kallio
Erkki Ojala
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Nokia Solutions and Networks Oy
Original Assignee
Nokia Siemens Networks Oy
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Nokia Siemens Networks Oy filed Critical Nokia Siemens Networks Oy
Priority to PCT/EP2008/050656 priority Critical patent/WO2009092437A1/fr
Publication of WO2009092437A1 publication Critical patent/WO2009092437A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1083In-session procedures
    • H04L65/1086In-session procedures session scope modification
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1083In-session procedures
    • H04L65/1095Inter-network session transfer or sharing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/12Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
    • H04M7/1205Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • H04M7/125Details of gateway equipment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/1016IP multimedia subsystem [IMS]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2201/00Electronic components, circuits, software, systems or apparatus used in telephone systems
    • H04M2201/52Telephonic communication in combination with fax communication
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2201/00Electronic components, circuits, software, systems or apparatus used in telephone systems
    • H04M2201/60Medium conversion

Definitions

  • the invention relates to a method, a session control en ⁇ tity, and a computer program product for modifying a speech call to a fax or data call in a communication network .
  • IP Internet Protocol
  • Analogue modem data and es- pecially facsimile telephony is still assumed to be im ⁇ portant in the future due to the fact that in existing fixed telecom equipments are connected to an IP Multime ⁇ dia Subsystem (IMS) defined by 3rd Generation Partnership Project (3GPP) via intelligent access device (IAD) or analogue telephone adapter (ATA) like adapter equip ⁇ ment.
  • IMS IP Multime ⁇ dia Subsystem
  • 3GPP 3rd Generation Partnership Project
  • IAD intelligent access device
  • ATA analogue telephone adapter
  • An ATA is an adapter that is used to connect a standard telephone to a high-speed modem to facilitate voice over IP (VoIP) and fax calls over the Internet.
  • facsimile has special legal status as for ⁇ mal undersigning or signature in some countries, which therefore need to be offered by service providers.
  • facsimile service is expected to be needed also in IMS networks that are Session Initiation Protocol (SIP) based and provide telephony services also for fixed subscribers.
  • SIP Session Initiation Protocol
  • IETF Internet Engineer ⁇ ing Task Force
  • CSCF call state control function
  • a media gateway control function acts as an in- terworking point between a circuit switched (CS) network and an IP network in the control plane of the network.
  • the MGCF controls the parts of the call state related to connection control for media channels in a media gateway (MGW) , communicates with call state control, and per ⁇ forms protocol conversion between the call control pro ⁇ tocols, such as SIP and integrated services digital net ⁇ work (ISDN) user part (ISUP) .
  • ISUP is a call control component of the signaling system No. 7 (SS7) protocol that determines the procedures for setting up, coordi ⁇ nating, and taking down trunk calls in the SS7 network.
  • a network element acting as MGCF is responsible for sig ⁇ nalling level interworking between CS and IMS domains.
  • a mobile switching center (MSC) is a mobile network ele- ment in the CS domain which performs the switching func ⁇ tions in its area of operation and controls the inter ⁇ working with other networks.
  • a home location register (HLR) is a register of a CS mobile network where all subscriber parameters of a mobile subscriber are perma- nently stored.
  • ETSI European Telecommunications Standards Institute
  • TISPAN Internet converged Services and Protocols for Advanced Networking
  • ITU International Telecommunication Union
  • ITU-T International Telecommunication Union
  • ITU-T V.152 Real time facsimile service
  • All mechanisms are supported in varying degree within different equipments and solu ⁇ tions.
  • Importance of T.38 real-time facsimile has been recently raised due to the fact that it will guarantee sending and receiving of facsimile documents in poor quality of service (QoS) conditions, which is clear benefit over other possible mechanisms.
  • QoS quality of service
  • Analogue data similarly also can use ITU-T V.152, ITU-T V.151 (Modem over IP) or even in-band transfer of data over G.711 codec. So far analogue fac ⁇ simile and modem services have not been included in the IMS by the 3GPP.
  • the object of the invention is to overcome the above drawbacks .
  • the invention solves the problem of prior art by provid ⁇ ing a method and a session control entity, comprising controlling a first session comprising a first media type towards a first user by means of a control protocol of a packet switched network, controlling the first ses ⁇ sion towards a second user by means of a control proto- col of a circuit switched network, receiving a request from the first user to modify the first session by changing media type from the first media to a second me ⁇ dia type, releasing the first session towards the second user only, establishing a new session comprising the second media type towards the second user, modifying the first session towards the first user by changing the me ⁇ dia type from the first media type to the second media type, and associating the new session towards the first user with the modified first session towards the second user.
  • the method or session control entity may further include controlling conversion of the first media type and the second media type between the packet switched network and the circuit switched network.
  • the controlling the conversion of the first media type and the second media type may comprise controlling a media gateway entity.
  • the method or session control entity may further com ⁇ prise charging the first session and the new session, and may comprise charging the first session and the new session as a single session.
  • the control protocol of the packet switched network may be a session initiation pro ⁇ tocol (SIP) .
  • the control protocol of the circuit switched network may be an integrated services digital network (ISDN) user part (ISUP) .
  • the request to modify the first session may be a re-INVITE or an UPDATE request of a session initiation protocol (SIP) .
  • the first media type may be for example a speech call, a facsimile or an analogue data call
  • the second media type may be another one of the speech call, the facsimile and the analogue data call.
  • the request to modify the first session may be a request to change the media type to T.38 fax.
  • Establishing the new session towards the sec- ond user may comprise indicating group 3 fax in a high layer compatibility (HLC) parameter of an initial ad ⁇ dress message (IAM) of integrated services digital net ⁇ work (ISDN) user part (ISUP) .
  • HLC high layer compatibility
  • IAM initial ad ⁇ dress message
  • ISDN integrated services digital net ⁇ work
  • ISUP integrated services digital net ⁇ work
  • controlling the first session may comprise establishing the session originating from the packet switched network and terminating to the circuit switched network, or establishing the session originating from the circuit switched net- work and terminating to the packet switched network.
  • the present invention has the advantage that it provides a solution for routing a call originated by IMS sub ⁇ scriber as a voice-only session and dynamically upgrad- ing the call into corresponding data call without re ⁇ initiating the call by end users.
  • Figure 1 presents an overview of prior art network architecture relevant for this invention.
  • Figure 2 presents a signaling flow of an embodiment of the invention.
  • FIG. 3 illustrates an internal structure and functions of a session control element implementing aspects of the invention .
  • FIG. 1 presents an overview of prior art network architecture relevant for this invention.
  • a first terminal device 1 is located in a packet switched communication network 8.
  • the first terminal device 1 may be a mobile or fixed line terminal and supports communication using at least one packet switched control protocol, such as SIP, for establishing and controlling communication ses- sions 11.
  • the packet switched communication network 8 may be for example, an IMS network or another internet or SIP based network, and a user of the first terminal device 1 may be a subscriber to the communication network 8.
  • a second terminal device 7 is located in a circuit switched communication network 9.
  • the second terminal device 7 may be a mobile or fixed line terminal and sup ⁇ ports communication using circuit switched control pro- tocol, such as ISUP, for establishing and controlling communication sessions 11.
  • the circuit switched communi ⁇ cation network 9 may be for example, a public switched telephone network (PSTN) or a public land mobile network (PLMN) , and a user of the second terminal device 7 may be a subscriber to the communication network 9.
  • PSTN public switched telephone network
  • PLMN public land mobile network
  • An MGCF 3 is interfacing the packet switched communica ⁇ tion network 8 and the circuit switched communication network 9, and is implementing protocol conversion be- tween control protocols of the communication networks 8 and 9, for example between SIP and ISUP, so that it is possible to establish communication 11 between the first terminal device 1 and the second terminal device 7 even though the first terminal device 1 and the second termi- nal device 7 do not support the same (single) protocol for controlling the communication 11.
  • the MGCF 3 is further communicating with a media gateway (MGW) 10, using an appropriate media gateway control protocol (MEGACO, MGCP) or H.248 protocol specified by ITU-T.
  • MGW media gateway
  • the user plane is a layer in which user data (e.g. modem or fax data) or speech coming from e.g. a radio network controller is converted to pulse code modulation (PCM) user data and vice versa. Tasks of the user plane are e.g. processing transcoding of speech channels and data format conversion for data.
  • the user plane traffic 12 associated with control plane communication 11 is routed through the MGW 10 between the packet switched communication network 8 and the circuit switched commu ⁇ nication network 9. Only the network elements relevant for interworking between CS and PS session are shown in Fig 1. In practice there are many other network elements (e.g. MSC, CSCF) involved in both control path 11 and user plane 12 but those are omitted for simplicity.
  • a facsimile or modem call may be initiated so that it is immediately known or recognized to be a data call.
  • a session description protocol (SDP) within an INVITE message of SIP protocol may contain directly information about the nature of the call e.g. about T.38, and no speech codec.
  • a facsimile call may also be initiated at first by a voice call (using e.g. G.729 codec) but then after a called party has answered, the call may be modi- fied to be the facsimile call by sending re-INVITE re ⁇ quest with a new session description.
  • voice call using e.g. G.729 codec
  • facsimile T.38
  • T.38 voice and facsimile
  • UE User equipment
  • SDP offer for example request for T.38 transfer.
  • the called subscriber may be a circuit switched subscriber and the call is routed to an MGCF by using standardized IMS procedures.
  • the MGCF examines the received SDP offer and if it supports re ⁇ quested media type (e.g. T.38) then the MGCF will con- tinue a normal call establishment towards the CS net ⁇ work.
  • the MGCF in this case sends an Initial Address Message (IAM) with Transmission Medium Requirement (TMR) parameter indicating 3.1 kHz audio as well as High Layer Compatibility (HLC) parameter encoded with information stating that the call is Group 3 Fax call. Also called party' s address (B number) is sent by the MGCF towards CS network.
  • the CS network may use either both the called party number (B number) information or received HLC parameter to determine that the call is a facsimile or an analogue modem call.
  • IP Multimedia media Gateway IM-MGW) handles required interworking in user plane (media) between the IMS and the CS domains.
  • Basic service group Basic service (1) (2)
  • TS Teleservice BS: Bearer service.
  • TS Teleservice BS: Bearer service.
  • the transparent facsimile services apply to A/Gb mode of GERAN only.
  • the non-transparent facsimile services apply to UTRAN only.
  • NOTE 4 The Voice Group Services, TS 91 and TS 92 apply to A/Gb mode of GERAN only.
  • the above table shows basic service groups defined by the 3GPP in the specification TS 22.004 "General on sup ⁇ plementary services”.
  • a model is provided that may be implemented and standardized to support also calls or sessions that begin by using a voice codec (speech call) but are afterwards switched into e.g. T.38 real-time facsimile call.
  • voice codec speech call
  • An example call flow is seen in Figure 2.
  • UE 1 of a calling IMS user sends an initial INVITE 201 request indicating only voice codecs in an SDP offer and the call is routed (possible via call state control function CSCF 2) to an MGCF 3. Then normal IMS-CS inter- working procedures apply so far.
  • the MGCF 3 is interfac ⁇ ing a packet switched domain (SIP UE 1, CSCF 2) and a circuit switched domain (gateway-MSC 4, HLR 5, Visited- MSC (VMSC) 6, CS UE 7) and based on the information re ⁇ ceived in the INVITE request, the MGCF 3 sends an IAM 202 message indicating 3.1 kHz audio call towards the called party.
  • the gateway-MSC 4 after receiving the IAM 202 message, queries 203 the HLR 5.
  • the HLR enquiry 203 is resulting the selection of TSIl service (speech) .
  • the gateway-MSC 4 forwards the IAM 204 to the VMSC 6 which completes the call signaling to the CS UE 7 in step 205.
  • the call is also completed at user plane 206.
  • part of the message flow is not shown, e.g. the CS/PS answer message flow from the CS UE 7 towards the SIP UE 1.
  • addi ⁇ tional network elements involved in the signaling path e.g. further CSCFs, border gateway control function (BGCF) etc.
  • the called party (CS UE 7) is a mobile subscriber, however, the called party may equally well be a fixed line subscriber, in which case the mobile specific parts of the CS network (gateway-MSC 4, HLR 5, VMSC 6) are to be replaced by corresponding fixed network elements, e.g. by telecommu ⁇ nication switches.
  • the SIP UE 1 sends a re-
  • INVITE 207 request to the same SIP dialogue that has al ⁇ ready been established towards the CS UE 7.
  • the MGCF 3 after receiving the re-INVITE 207, detects is step 208 that a media type (here fax) in the re-INVITE is differ- ent from the originally negotiated (here speech) and thereby generates a normal call clearing in release 209 message towards the CS UE 7 and a re-establishment of a new facsimile call towards the CS UE 7 with a new IAM 211 by using HLC indicating Group 3 Fax and using the same called party address that was used in the original call attempt, i.e. INVITE 201.
  • step 212 the HLR en ⁇ quiry results in selection of TS61 service (facsimile) .
  • Steps 213 and 214 complete the call establishment for the new call leg and corresponds to steps 204 and 205.
  • the CS UE 7 responds towards the SIP UE 1 by sending CONNECT message which is transmitted to the MGCF 3 in ISUP as ANSWER or CONNECT message in step 216.
  • the MGCF 3 may then determine that the establishment of the new call leg to the CS UE 7 was successful and acknowl- edges the re-INVITE 207 with a 200 OK SIP response in step 217.
  • the response may include information confirm ⁇ ing the change of the media or the service, e.g.
  • T.38 indication This way the CS UE 7 see the incoming fax call as a new call coming from the same calling party, however from SIP UE 1 point of view the same session has been modified to facsimile call.
  • the MGCF 3 handles the logic on behalf of the IMS subscriber.
  • a new capability for the MGCF is introduced to modify a CS side connec ⁇ tion by clearing an old speech connection and establishing a new facsimile call in order to create better end user experience for the IMS subscriber.
  • This procedure may be repeated more than one time during the same IMS session, meaning that the IMS subscriber could after a facsimile transfer continue the call as a voice call again, which require that the MGCF clears the fax call and re-establishes a new speech call towards the CS sub- scriber.
  • the procedure does not use alternative fax & speech service (TS61) but may be used and deployed with more traditional facsimile services (e.g. G3 fax) avail ⁇ able in today's CS mobile networks.
  • a timer is introduced in an MGCF and the MGCF is trying to establish a new call leg only after a certain time has expired. This may be required because termination of the first connection to ⁇ wards the CS UE may take time, meaning that if the MGCF would establish the new session immediately, the MGCF may get busy tone.
  • An SIP UE may try to update or upgrade a session, no matter which party initiated the session. This means that even if a CS UE has initiated the call/session, the IMS party may send a re-INVITE to swap from speech to fax or vice versa, and an MGCF may release the existing CS call and try to establish a new fax call to the CS UE.
  • both parties end users
  • the charged party may then change.
  • the situation (e.g. which party initiated the original session) may be indicated in the charging data records (CDR) and thereby the originally calling party may be charged for the whole call/session.
  • FIG. 3 illustrates the internal structure and func ⁇ tions of a session control element, e.g. MGCF, imple ⁇ menting this invention.
  • the MGCF 3 includes first ses ⁇ sion control unit 31 configured to control a call/session with a first media type (e.g. speech) to- wards a SIP UE 1.
  • the first session control unit 31 may control the call/session by means of a control protocol of a packet switched network, such as SIP.
  • a second ses ⁇ sion control unit 32 is configured to control the call/session towards a CS UE 7.
  • the second session con- trol unit 32 may control the call/session by means of a control protocol of a circuit switched network, such as ISUP.
  • the first and the second session control units 31 and 32 may control both originating and terminating sessions.
  • a receiving unit 33 is configured to receive, during the active call/session, a request to modify the call/session by changing the media type or a service from the current media/service (e.g. speech, facsimile, analog data call) to another media/service (e.g. speech, facsimile, analog data call) .
  • the receiving unit 33 may receive the request in a SIP re-INVITE or in a SIP UPDATE requests.
  • the receiving unit 33 may first check with the session control units 31, 32 which media or service is currently used.
  • the receiving unit 33 and the session control units 31, 32 may determine whether change of media or service is possible for the call/session.
  • a releasing unit 34 may be configured to release the call/session towards the CS UE 7 only if the change of media or service is supported.
  • the releasing unit 34 may release the call/session by sending a REL message of ISUP.
  • An establishing unit 35 is configured to establish a new call/session with the other media type or service towards the CS UE 7.
  • a modifying unit 36 is configured to modify the remaining part of the origi- nal call/session towards the SIP UE 1.
  • the modifying unit 36 may change the media type or service from the original media type or service (controlled by the first session control unit 31) to the other media type or ser ⁇ vice indicated in the request received by the receiving unit 33.
  • the modifying unit 36 may send an indication of the changed/new media type or service to the SIP UE 1 in a 200 OK SIP response.
  • An associating unit 37 is configured to associate the new call/session towards the CS UE 7 with the modified call/session towards the SIP UE 1.
  • a controlling unit 38 may control conversion of a media stream of the first media and/or a media stream of the second media between the packet switched network and the circuit switched network.
  • the controlling unit 38 may control a media gateway entity 10 through which the stream of the first and the second media type traverse at a user plane.
  • the controlling may be implemented us ⁇ ing for example MGCP, MEGACO or H.248 protocol.
  • the MGCF 3 may have a charging unit 39a that is collecting charging related information for communica- tion controlled by the MGCF 3.
  • the charging unit 39a may charge the first session and the second session as a single session. If the first session is initiated by the CS UE 7, a user of the CS UE 7 may be charged for the whole session if the SIP UE 1 initiates the change of media type and the new call/session towards the CS UE 7 is established by the establishing unit 35.
  • a charged party may be changed by the charging unit 39a when the type of media is changed so that the party initiating the change of media type will be charged.
  • Charging tar ⁇ iff or rate may be changed according to a media type in use.
  • a monitoring unit 39b may be configured to monitor that the CS connection is fully released before the es ⁇ tablishing unit 35 is trying to establish the new call/session.
  • the monitoring unit 39b may be implemented as a timer and the establishing unit 35 may start estab ⁇ lishing the new connection only after the timer has expired.
  • All units described above may be implemented e.g using microprocessors and/or other electrical components and/or by software.
  • the invention is not limited to mobile networks, IMS networks and to interworking of them, but may also be applied in other type of networks having similar session gateway entity role as MGCF and supporting plurality of communication services to enable switching from one service to another service during the session. Therefore, the MGCF is only used here as an example of a session control entity.
  • Functions of the session control entity (MGCF) described above may be implemented by code means, as software, and loaded into memory of a computer.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Multimedia (AREA)
  • Telephonic Communication Services (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

La présente invention concerne une MGCF (Media Gateway Control Function) modifiant un appel vocal en un appel de télécopie. La MGCF contrôle une première session comprenant un premier type de support vers un premier utilisateur au moyen d'un protocole de contrôle d'un réseau à commutation de paquets et vers un second utilisateur au moyen d'un protocole de contrôle d'un réseau à commutation de circuits. La MGCF reçoit de la part du premier utilisateur une demande de modification de la première session par le changement du type de support du premier support en un second type de support. La MGCF libère uniquement la première session vers le second utilisateur et établit une nouvelle session comprenant le second type de support vers le second utilisateur et modifie la première session vers le premier utilisateur en changeant le type de support du premier type de support au second type de support. La MGCF associe la nouvelle session vers le premier utilisateur à la première session modifiée vers le second utilisateur.
PCT/EP2008/050656 2008-01-21 2008-01-21 Réalisation d'un appel de données de télécopie et analogiques Ceased WO2009092437A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
PCT/EP2008/050656 WO2009092437A1 (fr) 2008-01-21 2008-01-21 Réalisation d'un appel de données de télécopie et analogiques

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
PCT/EP2008/050656 WO2009092437A1 (fr) 2008-01-21 2008-01-21 Réalisation d'un appel de données de télécopie et analogiques

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WO2009092437A1 true WO2009092437A1 (fr) 2009-07-30

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103731898A (zh) * 2012-10-12 2014-04-16 北京信威通信技术股份有限公司 一种路由sip呼叫请求的方法
CN107251512A (zh) * 2015-01-21 2017-10-13 瑞典爱立信有限公司 与安全媒体的ims会话的增强建立
US11290685B2 (en) * 2013-07-03 2022-03-29 Huawei Technolgoies Co., Ltd. Call processing method and gateway

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2002052811A1 (fr) * 2000-12-22 2002-07-04 Nokia Corporation Procede et systeme de modification d'un parametre de connexion
WO2004045182A1 (fr) * 2002-11-14 2004-05-27 Siemens Aktiengesellschaft Transmission de parametres de commande d'appel entre deux controleurs de passerelle de telecommunication dans des reseaux sip/sip-t

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2002052811A1 (fr) * 2000-12-22 2002-07-04 Nokia Corporation Procede et systeme de modification d'un parametre de connexion
WO2004045182A1 (fr) * 2002-11-14 2004-05-27 Siemens Aktiengesellschaft Transmission de parametres de commande d'appel entre deux controleurs de passerelle de telecommunication dans des reseaux sip/sip-t

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103731898A (zh) * 2012-10-12 2014-04-16 北京信威通信技术股份有限公司 一种路由sip呼叫请求的方法
US11290685B2 (en) * 2013-07-03 2022-03-29 Huawei Technolgoies Co., Ltd. Call processing method and gateway
CN107251512A (zh) * 2015-01-21 2017-10-13 瑞典爱立信有限公司 与安全媒体的ims会话的增强建立

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