Volte Using Ims

Voice over LTE (VoLTE) Using IMS

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3 | Voice over LTE (VoLTE) Using IMS

LTE_710c Version 1.1c 1


Award Solutions Proprietary - Do Not Copy2


f S SReferences:

3GPP - www.3gpp.org/ftp/Specs/html-info:

1. TS 23.203 Policy and charging controlarchitecture

2. TS 23.882 3GPP System Architecture Evolution

3. TS 23.237 IP Multimedia Subsystem (IMS) Service

Networks, Nokia, Samsung and Sony Ericsson,November 2010 One Voice; Voice over IMS Profile

3. TS 23.237 IP Multimedia Subsystem (IMS) ServiceContinuity

4. TS 23.292 IP Multimedia Subsystem (IMS)centralized services

5. TS 23.507 Protocols for Advanced Networking(TISPAN) Voice Call Continuity (VCC)

6 TS 29 213 Policy and Charging Control Signaling6. TS 29.213 Policy and Charging Control Signalingflows and QoS parameter mapping

IETF - www.ietf.org/rfc.html:

1. RFC 3588 Diameter Base Protocol

Others:

1. AT&T, Orange, Telefonica, TeliaSonera, Verizon,Vodafone, Alcatel-Lucent, Ericsson, Nokia Siemens

Award Solutions Proprietary - Do Not Copy 3


Voice over LTE (VoLTE) specifies how to support VoIPusing LTE. One Voice Profile, which is accepted by theVoLTE initiative, defines a narrow set of functions andoptions to support VoIP calls using the IMS and LTE. Theidea is to remove much of the confusion that comes withthe immense number of options that are inherent in theIMS. The simplification that comes with reduced optionslowers market risks Hence the LTE air interface protocollowers market risks. Hence, the LTE air interface protocolstack is optimized for VoIP performance. For example, thePDCP layer of the air interface protocol stack is obliged toimplement Robust Header Compression (RoHC).Additionally, suitable EPS bearers have to be established.

LTE supports nine levels of QoS. Hence, QCIs with propercharacteristics should be employed to support VoIPsignaling (QCI=5) and VoIP user traffic (QCI=1). While thediagram shows TCP/IP for the signaling operations, themodel is also valid for UDP/IP as appropriate for VoIP usertraffic. SIP is used as the signaling protocol to set up andmanage a VoIP call. The HTTP noted in the diagramincludes Extensible Markup Language (XML) ConfigurationAccess Protocol (XCAP) as well. The use of the AMR codecis requiredis required.



The key characteristics of the voice-medium and relatedfeatures specified in VoLTE are as follows.

Medium: All eight modes of the AMR codec aresupported. For example, 12.2 kbps AMR mode couldbe a widely used mode. Source rate controlledoperation means that the actual data rate couldchange within the call based on the source. To carry

accept an RTP packet that has up to 12 speechframes (e.g., redundant frames to improve reliability),one speech frame per RTP packet should berequested by the UE. In other words, the ptimeattribute is set to 20 and the maxptime attribute isset to 240 during the SDP negotiation. The UE usesthe same port number of RTP/UDP to send andreceive RTP packets RTCP is not generally used while

such an AMR payload, two formats are supported:bandwidth efficient and octet aligned. During the callset up, such negotiation of the AMR payload formatwould occur. For a given VoIP session, one of the twopayload formats is chosen. In the octet-aligned formatof the payload, all the fields in a payload, includingthe payload header, table of contents entries and

receive RTP packets. RTCP is not generally used whilespeech is on. Only when the media is on hold, isRTCP used. SDP capability negotiation framework(i.e., SDPCapNeg) is not used in the SDP offer butrather the RTP profile AVP IETF defined in RFC 3551is used.

speech frames themselves, are individually aligned tooctet boundaries for efficient implementation. In thebandwidth-efficient format, only the full payload isoctet aligned, resulting in fewer padding bits.

IP-based Protocols: To reduce signaling overhead,RoHC is used at the PDCP layer of the air interfaceprotocol stack. The AMR payload is placed into anRTP packet. While the receiver should be able to



VoLTE relies upon Session Initiation Protocol (SIP) forsignaling. The UE and the network exchange SIP signalingmessages to set up, manage, and tear down a VoIP call. Infact, the IMS uses SIP as the control protocol. An auxiliaryprotocol called Session Description Protocol (SDP) isembedded in a SIP message to facilitate QoS negotiation.

SDP defines characteristics of the media. For example, inthe case of a VoIP call, SDP may define Adaptive MultiRate (AMR) as the speech codec with 12.2 kbps data rate.VoLTE supports certain QoS preconditions. The basic SDPoffer/answer model is exploited. When the IMS session isoriginated (a VoIP call origination), the precondition-tagis included. More specifically, the Supported header isused, not the Require header, to convey the support forsuch QoS preconditions. While VoLTE requires the UE tosupport QoS preconditions, the operator can choosewhether to use this feature in the network or not.



The diagram shows a simplified view of the IMSarchitecture as defined by 3GPP. It is a rendering of themore general NGN architecture. IMS offers both generalpacket data support and multimedia session capabilities.The multimedia session capabilities are built on top of thegeneral packet data support capabilities. The generalpacket data capabilities may be deployed without themultimedia session capabilities

A second role of the CSCF is that of the primary callprocessing server. IMS calls this the Serving CSCF (S-CSCF). A UE registers with a specific S-CSCF. That S-CSCFprocesses all the IMS requests from the UE and itprovides access to requested services in the IMS servicesnetwork, i.e., connection to a certain Application Server(AS). The other IMS network elements displayed in thisslide jointly define a gateway function enabling themultimedia session capabilities.

Some network entities (the HSS for instance) may becommon to the EPC and the IMS network. Note also thatthe CSCFs provide access to a separate services networkthrough Application Server (AS) scenarios. This is one ofthe primary benefits offered by the IMS, and provides aplatform for the introduction of creative new services

slide jointly define a gateway function enabling theinterconnection (interworking) between IMS and variousexternal networks. For example, a voice call initiated aspacket-based VoIP in IMS may connect to a circuitswitched phone in the PSTN. The IMS gateway willfacilitate the call setup and the real-time translation of themedia between the packet-switched and circuit-switcheddomains

without the need to integrate these services into thetransport network.

The CSCFs play various roles in an IMS network, and IMStags each role with a different name. The Proxy CSCF (P-CSCF) serves as the entry point to the IMS network. DuringIMS registration, an IPSec tunnel is established betweenthe UE and the P-CSCF to assure data integrity.

domains.



The Proxy CSCF (P-CSCF) serves as the users outboundproxy server. In this role the P-CSCFs principalresponsibility is to proxy messages between the user andthe Serving CSCF (S-CSCF), but it performs some otherfunctions as well.

As the users first point of contact within the IMS network,the P-CSCF plays an important role in detectingemergency calls and routing them to the proper location.The P-CSCF can also implement the operators securitypolicy by establishing secure tunnels to the user or toother roaming partners. Since the P-CSCF is the IMSnetwork element closest to the access network, it canalso play a role in enforcing Quality of Service (QoS) policy.If the operator wishes to conserve bandwidth on theaccess network (always a big concern for wirelessnetworks), the P-CSCF can perform SIP compression.



The Interrogating CSCF (I-CSCF) is the first point of contactwithin the users home network. Its a stateless machinewhose main job is to query the HSS and find the locationof the S-CSCF.

The I-CSCF is an optional node in the IMS architecture; theP-CSCF can be configured to contact the S-CSCF directly.

The I-CSCF has a number of functions It may perform loadThe I-CSCF has a number of functions. It may perform loadbalancing between the S-CSCFs with the support of theHSS. The I-CSCF may also hide the specific configurationof the home network by providing a single point of entryinto and an exit from the home network. To perform thisfunction, the I-CSCF applies topology hiding to SIP headersthat would otherwise reveal the topology of the homenetwork (e.g., Via and Route entries). The topology hiding( g , ) p gy gfeature is accomplished by encrypting and decrypting thetopologically sensitive information. The handling ofrequests and responses are independent of each other sothat no state information for a session is required to bekept in the I-CSCF. The I-CSCF may also perform someforms of billing and support a firewall function.



The Serving Call Session Control Function (S-CSCF) is themain node that performs session management for IMS.There can be several S-CSCFs in the network. They can beadded as needed based on the capability requirementsand/or the capacity requirements of the network. The S-CSCF in the home network is responsible for all sessioncontrol tasks. The S-CSCF also stores the users serviceprofile downloaded from the HSS that is it plays the roleprofile downloaded from the HSS, that is, it plays the roleof the Session Initiation Protocol (SIP) Registrar. The S-CSCF is chosen based on its capability to service anythingthe subscriber is authorized to do. The definition of thesecapabilities is not standardized and, therefore, is left up tothe operator.



Among the IMS database building blocks, the HomeSubscriber Server (HSS) is the key element.

As in the legacy mobile network, a centralized subscriberdatabase is still needed in the IMS. The Home LocationRegister (HLR) has evolved into the Home SubscriberServer (HSS) with the inclusion of IMS profiles andrecords. Diameter-based interfaces have been added aswell. The HSS interfaces with the I-CSCF and the S-CSCF inorder to provide information about the location of thesubscriber and the subscribers subscription information.The HSS and the CSCF communicate via the Diameter-based Cx interface. The HSS also communicates with theApplication Servers (ASs) in the services network througheither the Sh or the Si interfaces.

Although not specifically included in this slide, aSubscription Location Function (SLF) can also be includedunder the database category. The SLF is an optionalnetwork element with a very narrow responsibility. It isduring the registration process that the SLF might have todetermine which HSS has the subscribers profile in largeIMS networks in which there may be several HSSsdeployed.



Several gateways fall under the category of theInterworking Building Block. This kind combined gatewayis made up of a Media Gateway Control Function (MGCF),a Media Gateway (MGW), and a Signaling Gateway (SGW).The MGCF controls one or more MGWs, which allows formore scalability in the network. The MGCF manages themedia connection between the Public Switched TelephoneNetwork (PSTN) bearer (the trunk) and the users IP media

assign for media translation. Using a protocol calledMEGACO, the MGCF establishes a context within theselected MGW. Through a series of transactions andreplies, the MGCF informs the MGW how to establish thecontext.

The MGCF also translates the INVITE message into anequivalent ISUP message - an Initial Address Message

Network (PSTN) bearer (the trunk) and the user s IP mediastream. This connection is referred to as a context. Inaddition, the MGCF translates the Session InitiationProtocol (SIP) call setup messages used in the IP domaininto SS7 call setup messages used in the PSTN. TheMGCF and MGW can be distributed as demonstrated inthis slide, or the two functions may be co-located, in whichcase they are jointly referred to as an integrated

(IAM) is one example. The MGCF sends the ISUP message,via IP, to the SGW. The MGCF is a software or serverfunction. This means it must run on a highly availablecomputing platform, not on a real-time hardwareprocessing environment.

case they are jointly referred to as an integratedgateway.

The MGCF converts SIP messages into ISDN SignalingUser Part (ISUP) messages. These messages areforwarded to the SS7 network. After the MGCF receivesthe SIP INVITE message from the Call Session ControlFunction (CSCF), or the Breakout Gateway ControlFunction (BGCF) it determines which of its MGWs toFunction (BGCF), it determines which of its MGWs to



The PSTN currently lives exclusively in the SS7 world; itonly understands SS7-based signaling, and there is noincentive for it to provide support for anything other thanSS7. SS7 is not as flexible as IP. Furthermore, it is notdeployed in a public environment. SS7 is only deployed inprivate networks, or within an operators network tosupport call control and mobility management.

To avoid the need for the MGCF to support SS7, the IMSassumes the existence of a Signaling Gateway (SGW). TheSGWs job is to receive the ISUP message that is ridingover IP transport protocols, and convert the lower layers to(or from, if coming into the IP network from the PSTN)enable transport over the Message Transport Protocol(MTP), which is the native environment for SS7.

The example shown in this slide demonstrates theestablishment of a session between the IP environmentand the Public Switched Telephone Network (PSTN). Thisis when the MGCF converts the SIP INVITE message intoan ISUP IAM, and forwards that message to the SGW asan IP packet (i.e., IP transport). The SGW preserves theISUP IAM message, but converts the lower layers into MTPto transport it over the SS7 network.



If the Media Gateway Control Function (MGCF) is thebrains of the media operations, then the Media Gateway(MGW) is the brawn. It is the workhorse that performs theprocessing of the media bits between end users. Itsprimary function is to convert media from one format toanother. In wireless networks, this is predominantlybetween Pulse Code Modulation (PCM) in the PSTN andan IP based vocoder formatan IP-based vocoder format.

Under the direction of the MGCF, the MGW creates acontext between two terminations. A termination is aMegaco term for a bearer. The format is PCM for the PSTNbearer and IP packets for the IP-based vocoder stream.

The MGW is likely to be a real-time hardware-basedplatform. It is critical that it processes the bits as quicklyp p q yas possible so that delay is not added to the transmissionof the information.



There are three different types of Application Servers:

1. A local SIP Application Server (AS), where the AShosts and executes IMS SIP-based services; this maybe used to run new SIP-based applications.

2. The Open Service Access Service Capability Server(OSA-SCS) is a gateway entity that allows forapplication hosting external to the UMTS networkapplication hosting external to the UMTS network.The SCS serves as a placeholder that allows a call tobe controlled by an external server through anuntrusted network. The SCS terminates the ISC/SIPinterface, and then communicates with theapplication server through an API such as the ParlayMultimedia Call Control API. The SCS also maintainsany security associations with the external network.y y

3. The IP Multimedia Services Switching Function (IM-SSF) enables the UE to reuse legacy services basedon Intelligent Networks.



This figure summarizes the major control plane and userplane protocols used in the IMS. The network layerprotocol is IP and runs over various data-link and physicallayer protocols. Three protocols are used at the transportlayer. The Transmission Control Protocol (TCP) maytransport Session Initiation Protocol (SIP) messages. TheUser Datagram Protocol (UDP) is used to carry real-timetraffic such as Voice over IP (VoIP) Either a Stream Control

or more real-time multimedia flows. It is typically usedin conjunction with the RTP, which delivers the actualtraffic. An RTSP session may set up multiple TCPconnections for the reliable delivery of remote controlcommands, or it may use the UDP for the samepurpose.

Other control protocols used in IMS include SIP andtraffic such as Voice over IP (VoIP). Either a Stream ControlTransmission Protocol (SCTP) or TCP may be used to carryDiameter messages.

The major application-level signaling and traffic protocolsare also listed:

A real-time application may encapsulate mediasamples with the Real-time Transport Protocol (RTP),

the Session Description Protocol (SDP), which iscarried in the body of a SIP message, and Megaco (orITU H.248) protocol, as well as the Real-timeTransport Control Protocol (RTCP).

The Diameter-based protocol is used to exchangepolicy information between a Policy and ChargingControl Function (PCRF) and Policy and Chargingp p ( ),

which is, in turn, transported using the UDP.

The Real-time Transport Control Protocol (RTCP) doesnot transmit any media streams itself but providesstatistics and control information to meet the desiredQoS for RTP traffic.

The Real-time Transport Streaming Protocol (RTSP) is

( ) y g gEnforcement Function, or PCEFs. This reference point(Gx) uses SCTP as the transport protocol to exchangethese Diameter messages.

a control protocol used to establish and manage one



Lets walk through a very basic SIP session setup in whichAlice calls Bob. Here, Alice is the calling party and Bob isthe called party.

1. The calling party, referred to as the User Agent Client(UAC), sends the INVITE request to the called party,the User Agent Server (UAS). The invite requestcontains a session description that specifies the

subject of the call.

3. Once the called party decides to accept the call, theUAS sends a final response to the INVITE request. Itcontains a session description indicating the mediaformats that the called party wishes to receive. Notethat the two media formats do not need to be thesame.

media formats the caller wishes to receive, and othersession parameters such as the IP address and portnumber to which the called party should direct mediawhen the time comes.

2. The UAS may send zero or more provisionalresponses to indicate the progression of the call. TheTrying response indicates that it will try to establish

4. The UAC acknowledges that it received the finalINVITE response by sending the Acknowledgement(ACK) request.

5. The UAs may send in-call messaging for variouspurposes. Examples include the addition or removalof media from the call, addition of users, Call Hold,

t I h i t th INVITE gy g p y

the session. The Ringing response indicates that thecalled party has been located and the server will alertthe called user to the presence of an incoming call. Itmay be helpful to point out that the exact nature ofthe alert depends entirely on the user interfacecapabilities of the end-user device. A simpletelephone may ring, while a computer terminal may

etc. In such circumstances, the INVITE message canbe reissued.

6. Eventually, either the calling party (as in this example)or the called party will want to release the call. Torelease the call, a BYE request is sent.

7. Completing our example, the called party responds tothe BYE request with an OK response.

display a dialog box describing who is calling and thethe BYE request with an OK response.



SIP works in concert with the Session Description Protocol(SDP). SIP deals with the session-management issueswhile SDP is responsible for the session description. SDPmessages are transported in the body of SIP messages.This split between session management and sessiondescription is a powerful arrangement because it allowsthe two functions to evolve separately. For example, newmedia types can be easily incorporated by defining new

have a connection address.

The media description contains:

Media Type: Voice, Video, Text, Application, orMessage;

Port: Port number where the media is to be received;

Transport Protocol: Definition of the transport layermedia types can be easily incorporated by defining newsession descriptors without impacting SIP functionality atall.

RFC 4566 defines the purpose and syntax of SDP. Itspecifies that a multimedia session is a set of multimediasenders and receivers. It is not a true protocol but alanguage for representing the key parameters that

Transport Protocol: Definition of the transport layerprotocol to be used (e.g., RTP, RTP/AVP, UDP, TCP,MSRP/TCP, etc.);

Media Format: This is a list of the possible formats ofthe media (e.g., g.711, h.261, etc.);

Under each media description, a number of attributes canbe defined that further specify the media characteristics.

characterize a multimedia session.

The key elements of the SDP are described below.

The connection line contains information about theconnection address. This is the address where the SDPsender expects to receive the incoming media packets.This line can be present at the session so that all media isdirected to this address or each individual media line can

For example, one attribute defines directionality(send/receive, receive only, send only). Another attributeindicates the sender does not want to send or receivemedia (inactive). There are a number of attributes thatmay be defined this way.

directed to this address, or each individual media line can



The IP telephony application encapsulates its mediapackets into the Real-time Transport Protocol (RTP).Defined in RFC 3550, the RTP provides a commonframework that can be used to transport voice, video, orany other form of real-time media. The RTP, in turn, runson top of the User Datagram Protocol (UDP).

The RTP provides a general set of functions that mostreal-time applications need regardless of the type ofmedia: sequential numbering and time-stamping. Itprovides the capability on the receiving end to reassemblemedia packets in the correct order, play them back at theright times, and to identify lost packets. It can alsosynchronize multiple packet streams and identify thealgorithm that was used to encode the packet.



The Real-time Transport Protocol (RTP) often uses acompanion protocol, called the RTP Control Protocol(RTCP), to monitor the sessions Quality of Service (QoS).RTCP provides out-of-band statistics and controlinformation for an RTP flow. The RTCP defines messagessuch as the Sender Report (SR) and Receiver Report (RR).These messages allow the sender and receiver tomeasure QoS parameters such as packet loss average

203 Bye Message Terminates a traffic stream.

204 Application Specific Message Enables applications tocreate their own message.

Information exchanged through sender and receiverreports includes the:

Version of RTP being used,measure QoS parameters such as packet loss, averagedelay and jitter. The applications can deduce theperceived quality from these parameters. The RTCP isbased on the periodic transmission of control packets toall participants in the session, using the same distributionmechanism as the data packets.

RTCP message types:

Network Time Protocol (NTP) timestamp, and

RTP timestamp used for intra-/inter-mediasynchronization (RTP sends media streamsseparately).

200 Sender Report Provides an absolute timestamp toenable the synchronization of multiple mediastreams.

201 Receiver Report Informs the source about conditionsof reception. Enables the receiver to adapt tosenders transmit conditions.

202 Source Description Message



S S SSubscribers can have several identities in IMS.

The subscribers private identity is a globally unique,permanent identity assigned by the home networkoperator. Its used for registration, authorization,administration, and accounting purposes. Every IMS userhas at least one of these. The ID may be derived from anIMSI.

For local number addressing, IMS will use the dial-stringas provided by the UE for the user part of the SIP URI. Thephone context will be set to the home domain name.

The subscribers public identity is used for actualcommunications. These kinds of identities can bepublished on business cards or Web sites. There are likelyto be multiple IMPUs for an IMPI. The IMPU can be sharedamong phones and terminals so that either device can bereached with the same identity, a single phone-number foran entire family.y

Both the private and the public identities are not phonenumbers, but they are Uniform Resource Identifiers (URIs),that can be a string of digits (a Tel URI, like tel:+1-555-1231234) or alphanumeric identifiers (a SIP URI, likesip:[email protected]).



The IMS call setup procedure is a modified version of thestandard kind of SIP-based call setups. The chiefmodification is the need to ensure access resources areprovisioned before letting the phone ring.

Here are the relevant steps in the procedure:

Media Negotiation: The two endpoints must decidewhat types of media they will exchange (audio video

Ringing: Once all Quality of Service (QoS)preconditions have been satisfied and the accessresources have been established, the called partymay be alerted to the incoming call.

Answer: Once the called party accepts the call, thetwo endpoints begin exchanging media packets.

Media Flow: The two endpoints exchange voicewhat types of media they will exchange (audio, video,application data), in which directions it will be sent(send-only, send-and-receive), and how the mediapackets will be encoded and decoded.

QoS Negotiation: The two endpoints must statewhether they require dedicated resources to bereserved for the call and whether those resourcesh b f ll g t d b f th ll i

Media Flow: The two endpoints exchange voice,video, or application data packets. While not shownexplicitly on the slide, the media packets may be sentdirectly between the endpoints. They do not traversethe same path as the call signaling.

have been successfully granted before the call isallowed to proceed.

Resource Reservation: The two endpoints mayrequest that their local access network reservededicated resources for the media flows in this call.The local networks may, in turn, request that theusers home network authorize that request.



This figure illustrates a more elaborate IMS call flow, inwhich media negotiation is done using two offer/answerexchanges. This scenario assumes that the originatormakes the final codec selection, that both sides requireresource reservation, and neither side pre-allocatesresources but rather waits until the codecs have beendecided before requesting resources.

The first offer contains the desired media streams andlists two acceptable codecs (iLBC and AMR). It indicatesthat local resource reservation is required, but notcurrently allocated. The first answer accepts the mediastream and returns its acceptable codecs.

The originator selects AMR as the final codec and sends asecond offer. It subsequently requests the local accessq y qnetwork to allocate resources necessary for the AMRmedia stream. The called party receives the final decisionand begins requesting resources as well. It responds witha second answer.



When the originator receives confirmation that the localaccess network has allocated resources for the session, athird offer is sent indicating that the QoS preconditionshave been satisfied. This is achieved by sending anUPDATE message. Basically, once the UE has madereservation with its access network for the required QoS, itinforms the other end of the session of such a QoSreservation through the SDP in the UPDATE message Thereservation through the SDP in the UPDATE message. Themain QoS parameters (e.g., AMR rate) can also beconveyed in the SDP of the UPDATE message. Likewise,the called party receives works with its own accessnetwork for QoS, receives confirmation on the QoS fromits access network, and indicates its own preconditionshave been satisfied in the answer via the 200 OKresponse It also proceeds to alert the user and sends theresponse. It also proceeds to alert the user and sends the180 Ringing response. Resource reservation for cellularnetworks is quite important as scarcity of radio resourcesmay prevent a session from being set up.



The figure shows a high-level message flow for dynamicservice data flow establishment.

After the UE registers, authenticates, obtains an IPaddress and has a default bearer established, the usercan start various services applicable to the QoS definedfor the default bearer.

When the user decides to start a service that requires aWhen the user decides to start a service that requires adifferent QoS than the default EPS bearer, a new EPSbearer must be established to carry that traffic. In theexample shown, SIP signaling for the IMS is carried on thedefault bearer. SIP is used to establish a service.

The PCRF controls the establishment of the service andpasses the QoS policy decision to the P-GW. The P-GW will

i th PCC l d l th li i t threceive the PCC rules and apply the policies to the newSDF and dedicated bearer. It then communicates with theS-GW, which gets the MME involved. The MME will get theeNB involved in establishing the S5 and S1 bearers.

The eNB performs admission control to determine ifenough bandwidth is available for the service data flow. Ifadmission control is successful, it uses RRC procedures toestablish a radio bearer.



Here are the main types of radio resources needed for aVoIP call:

Signaling Radio Bearers (SRBs): Regular SRB1 andSRB2 that carry typical Access Stratum and Non-Access Stratum signaling messages are established.These are the same as non-VoIP SRBs.

Acknowledged Mode (AM) Data Radio Bearer (DRB) Acknowledged Mode (AM) Data Radio Bearer (DRB)with QCI 8/9: An EPS bearer with QCI 8/9 is neededto carry the IMS signaling (e.g., registration relatedsignaling). The associated radio bearer usesacknowledged-mode RLC for reliability.

Acknowledged Mode (AM) Data Radio Bearer (DRB)with QCI 5: An EPS bearer with QCI 5 is needed to

t l V IP ig li g ( g th SIP INVITEcarry actual VoIP signaling (e.g., the SIP INVITEmessage). The associated radio bearer usesacknowledged-mode RLC for reliability.

Unacknowledged Mode (AM) Data Radio Bearer(DRB) with QCI 1: An EPS bearer with QCI 1 wouldactually carry speech frames. The associated radiobearer uses unacknowledged-mode RLC.



S S f fThe Application Server (AS) in the slide can provideSupplementary Services to IMS subscribers by, forexample, modifying SIP messages: it could insert a CallWaiting (CW) indication in a SIP INVITE message. The S-CSCF brings the AS into a call by way of the SIP-based IMSService Control (ISC) interface. The ISC interfaceaddresses a sophisticated concept in the IMSarchitecture which includes the notion of a service such

notifications on changes for the users data. ApplicationServers use the Sh interface to retrieve and updatesubscriber profiles and shared database information.Examples are the location of a service, the name of acertain S-CSCF name for terminating a SIP session, theregistration information for the user, or reminders for awake-up call services. The interface is also used forreceiving notifications on modifications of the subscribedarchitecture, which includes the notion of a service such

as modifications to a predefined scenario by which a newfunction appears both from a technical or userexperience. The ISC depends on relatively simple SIP-based mechanisms in which elements from the sessioncontrol plane among the CSCFs interact with elementsfrom the service control plane (where the ApplicationServers live) in order to modify some kind of SIP signaling

receiving notifications on modifications of the subscribeddata in the HSS.

The Cx interface is a Diameter-based interface used by aCSCF and the HSS for location management and userauthentication procedures.

The AS sometimes needs information best obtained fromthe subscriber. The Ut interface in the slide lives betweenServers live) in order to modify some kind of SIP signaling

flow.

The blue interfaces are SIP-based: Mw and ISC.

The red interfaces are Diameter-based interfaces. One ofthese is the Sh interface defined in the IMS for servicelayer-based AAA-type functions. The Sh interface allows forthe download and update of transparent and non-

the UE and the AS so that the subscriber can manipulateservices: managing policies, manipulating groupmemberships, setting event triggers, etc. The slide showsthe green Ut interface which, in the IMS architecture, letsan IMS client manipulate the AS using XCAP. XCAP is anHTTP-based protocol that manipulates what XML callsdocuments. IMS clients interact with service logic in the

transparent user data, and it allows for request and send AS through Web-based user interfaces.



f S S fAmong the vast array of Supplementary Services, VoLTErequires only a few, most of which have been in theIntelligent Network (IN) for a long time and tend to betaken for grated. Supplementary Services aremodifications to calls that used to live in the MSC back inthe circuit-switched days of telephony. They live on in theIMS as services like the identification of a caller or calledparty privacy of identification call diversion call barring

model that is capable of delivering presence, location andall kinds of preference services that could enable newforms of communication services that take advantage ofIP multimedia.

party, privacy of identification, call diversion, call barring,hold, message waiting, and conference calls.

Supplementary Services are supported with MultimediaTelephony Service (MMTel), which is an IMS standardoffering converged, fixed and mobile real-time multimediacommunication using all kinds of media: voice, real-timevideo, text, file transfers, sharing pictures, and sharingaudio and video clips. MMTel adds and drops mediaduring a session. You can start with chat, add voice (VoIP),add another caller, add some video, insert an avatar for atemporarily absent caller, share media, transfer files, anddrop any of these without losing the session.

MMTel is one of several options among many in theoverall migration of moving away from traditional SS7, out-of-band, circuit-switched-based signaling, and into a new



The IMS is a service delivery platform and facilitatesoffering a variety of IP-based services. While VoIP isexpected to be a popular IMS application, SMS is anotherattractive application that can be handled using the IMS.In the case of applications such as VoIP calls, the CSCFdoes not get involved in any user traffic like speech. TheUE needs to implement the functions of an SM-over-IPsender and an SM over IP receiver Features such assender and an SM-over-IP receiver. Features such asstatus reports, delivery reports, and notification ofmemory availability are also supported. The IMS corenetwork performs the functions of an IP Short MessageGateway (IP-SM-GW). Both UE-originated and UE-terminated SMSs are supported. The actual SMS iscarried as a "vnd.3gpp.sms" payload in a SIP MESSAGErequest An AS can act as an IP-SM-GW For a receiver torequest. An AS can act as an IP SM GW. For a receiver toget the SMS, the receiver needs to perform the IMSregistration and indicate its capability to receive traditionalshort messages over the IMS network by providing a"+g.3gpp.smsip" parameter into the Contact header of theregistration request message.


Acronyms

1xEV-DO 1x Evolution for Data Optimized 3GPP Third Generation Partnership Project AAA Authentication, Authorization and Accounting ACK Acknowledge or Acknowledgement ACM Address Complete Message AMR Adaptive Multi-Rate ANM Answer Message API Application Program Interface AS Application Server AT Access Terminal AVP Attribute Value Pair BGCF Breakout Gateway Control Function CSCF Call Session Control Function CSFB Circuit-Switched Fallback CW Call Waiting DCCH Dedicated Control Channel DNS Domain Name System DRB Data Radio Bearer E-UTRAN Evolved Universal Terrestrial Radio Access Network or Evolved UMTS Terrestrial Radio

Access Network eNB Evolved Node B or E-UTRAN Node B EPC Evolved Packet Core EPS Evolved Packet System HLR Home Location Register HSS Home Subscriber Server HTTP HyperText Transfer Protocol I-CSCF Interrogating CSCF IAM Initial Address Message ICID IMS Charging ID IETF Internet Engineering Task Force IM Idle Mode IMD IP Multimedia Domain IMPI IP Multimedia Private Identity IMPU IP Multimedia Public Identity IM-SSF IP Multimedia Services Switching Function IMS IP Multimedia Subsystem IMSI International Mobile Subscriber Identity IN Intelligent Networks IP Intelligent Peripheral IP-CAN IP Connectivity Access Network IPSec IP Security

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Acronyms

IPV4 Internet Protocol version 4 IPV6 Internet Protocol version 6 ISC IMS Service Control ISDN Integrated Services Digital Network ISUP ISDN Signaling User Part ITU International Telecommunication Union kbps kilo-bits per second LTE Long Term Evolution MEGACO Media Gateway Control MGCF Media Gateway Control Function MGW Media Gateway MME Mobility Management Entity MSC Mobile Switching Center MTP Message Transfer Part NGN Next Generation Network NTP Network Time Protocol OFDM Orthogonal Frequency Division Multiplexing OSA Open Service Access P-CSCF Proxy-CSCF P-GW Packet Data Network Gateway PCC Policy and Charging Control PCM Pulse Code Modulation PCRF Policy and Charging Rules Function PDCP Packet Data Convergence Protocol PDN Packet Data Node PRACK Provisional Response Acknowledgement PSTN Public Switched Telephone Network QCI QoS Class Identifier QoS Quality of Service RAN Radio Access Network RLC Radio Link Control RoHC Robust Header Compression RR Receiver Report RRC Radio Resource Control RTCP Real-Time Transport Control Protocol (or RTP Control Protocol) RTP Real-Time Transport Protocol RTSP Real-Time Streaming Protocol S-CSCF Serving CSCF S-GW Serving Gateway SC Single Carrier SCP Session Configuration Protocol

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Acronyms

SCP Service Control Point SCS Service Capability Server SCTP Stream Control Transmission Protocol SDF Service Data Flow SDP Session Description Protocol SIP Session Initiation Protocol SLF Subscription Location Function SMS Short Message Service SON Self-Organizing Network SR Sender Report SR-VCC Single-Radio Voice Call Continuity SS Supplementary Services SS7 Signaling System 7 SSF Service Switching Function TCP Transmission Control Protocol TCP/IP Transmission Control Protocol/Internet Protocol TD-LTE Time Division Long Term Evolution TS Technical Specification UAC User Agent Client UAS User Agent Server UDP User Datagram Protocol UE User Equipment UM Unacknowledged Mode UMTS Universal Mobile Telecommunications System URI Uniform Resource Identifier URL Uniform Resource Locator UTRAN Universal Terrestrial Radio Access Network or UMTS Terrestrial Radio Access Network VCC Voice Call Continuity VoIP Voice over Internet Protocol VoLTE Voice over LTE VPN Virtual Private Network WiMAX Worldwide Interoperability for Microwave Access XCAP XML Configuration Access Protocol XML Extensible Markup Language

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