Quinta Reunión Ministerial de Comercio - DEL AREA DE LIBRE COMERCIO DE LAS AMÉRICAS (ALCA)

CAPITULO II. DEL AREA DE LIBRE COMERCIO DE LAS AMÉRICAS (ALCA)

E. Quinta Reunión Ministerial de Comercio

SIP SIP

SIP (Session Initiation Protocol) is a signalling protocol with a number of functions.

Its first function is the initiation, modification and termination of interactive or multimedia communication sessions between two or more users over any IP network.

Because session initiation requires the user’s location to be determined, SIP supports user mobility and is responsible for finding the user’s location at the time a session is initiated. ‘Location’ in this sense means the user’s current IP address.

Once the user has been located, SIP ensures that both hosts involved in the session are informed of each other’s IP address; it also delivers a description of the session to which the terminating user is invited.

SIP is a signalling and discovery protocol and does not need to know details about the sessions it is helping to establish; that function is left to an associated protocol, SDP.

When the user has been located and the session description delivered, SIP conveys the response to the initiator, which could be an accept or a reject. If the invitation is accepted, the session becomes active.

LTE Evolved Packet Core Network

Sequence Number Timestamp Payload Type Synchronization ID

Voice RTP UDP IP

IP Network

RTP RTP

RTP is an application-layer framing protocol that enables real-time applications to stream data over an IP network. It defines a way to format IP packets carrying data isochronously (produced at regular constant intervals) and includes information on the payload type, timestamps and sequence numbers.

RTP is an end-to-end protocol, not a QoS protocol. It was designed to allow receivers to compensate for network-induced timing variations and mis-sequencing. It is typically used on top of UDP.

Each RTP packet carries a sequence number, timestamp and synchronization source identity. Depending on the application, these can be used in a number of ways.

A video application, for instance, can immediately deduce from the timestamp which part of the screen is described by the IP packet. An audio application uses the sequence number and timestamp to manage a reception buffer. If a packet does not arrive on time and is still missing when the information is due to be played back, the application may decide to copy the last packet played and repeat it long enough to catch up with the timestamp of the next received packet.

The synchronization source identity can be used, for example, by a receiver to synchronize audio and video streams during a video-conference.

Further Reading: IETF RFC1889/3550 (RTP & RTCP)

LT3604/v4.0 © Wray Castle Limited A.15

Delay since LSR RTCP Sender Reports Number of packets sent RTP timestamp

NTP timestamp (absolute time) Sender's packet count Sender's SSRC

Calculates the Round Trip Delay

( A – LSR – Delay Since Last Report) (i.e. Receiving time at receiver)

RTP Terminal

Terminal

RTCP RTCP

RTCP provides out-of-band control information for an RTP flow. It is used periodically to transmit control packets to participants in a multimedia session in order to provide feedback on the quality of that particular RTP session.

RTCP provides information such as bytes sent, packets sent, packets lost and jitter, as well as information that allows the round trip delay to be calculated to a particular receiver. An application may use this information to increase QoS, perhaps by limiting flow or by using a lower bit rate codec instead of a higher bit rate codec. RTCP may be considered to be providing QoS reporting.

There are several type of RTCP packet:

■ sender report packet

■ receiver report packet

■ source description RTCP packet

■ goodbye RTCP packet

■ application-specific RTCP packet

■ extended report packet

In theory, the use of RTCP is optional for RTP implementations, however in the LTE/VoLTE environment its use is mandatory. Inter-operable VoLTE standards are set out in the GSMA (GSM Association) IR.92 profile, which states (section 3.2.4) that RTCP must be supported by VoLTE UEs but that during ‘speech-only sessions’ RTCP reports must ‘speech-only be transmitted during periods when one or other of the parties is placed on hold. The RTCP flow therefore acts as a session keepalive during periods of inactivity.

An extension to RTCP, known as RTCP-XR (Extended Reporting) has been developed (specified in RFC3611) to meet the specific needs of VoIP transmission over RTP. RTCP-XR has an expanded report set that includes additional metrics that describe VoIP characteristics. These include packet loss and discard metrics, call and transmission quality metrics, jitter and delay metrics and other parameters that enable the quality of a VoIP session to be determined and reported.

Further Reading: IETF RFC1889/3550 (RTP & RTCP), RFC3611 (RTCP-XR), GSMA IR.92

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LTE Evolved Packet Core Network

Transport Address

Transport Address S1 – 10.10.1.100:2020

S1 – 10.47.21.38:4582 S1 – audio

Video component can be dropped without affecting the audio

S2 – 10.47.21.38:4583 S2 – 10.10.1.100:2021

S2 – video

RTP Sessions RTP Sessions

An RTP session is an association between a set of participants using RTP to communicate.

The example in the diagram shows two participants involved in a multimedia session.

In this case two RTP sessions are established, each with its own RTP and RTCP packet flows: session 1 (S1) is used for audio and session 2 (S2) for video.

Alternatively, an audio/video codec may multiplex the audio and video data onto a single RTP session.

3GPP suggests that separating the audio and video components of services like video calling may allow for a greater degree of resilience. If a UE is located near the edge of a cell or in any other area where high-speed coverage may not be guaranteed, the network may be able to drop the video part of the session but retain the audio in the event of the connection becoming unstable. This would allow the call to continue even if the video element was lost.

Note in this case that an RTP session may be identified by transport address pairs. A transport address is a combination of an IP address and port number. A transport address pair defines the IP address and port numbers used for both RTP and RTCP packets. It is standard procedure for the RTP port to be allocated an even-numbered port address while the corresponding RTCP port will be the next highest odd-numbered port.

Further Reading: IETF RFC1889/3550 (RTP & RTCP)

LT3604/v4.0 © Wray Castle Limited A.17

Appendix

IM Private Identity (IMPI)

IM Public Identity (IMPU)

Public Service Identity (PSI) NAI

identifies subscription, not user user@realm

SIP URL or ‘tel’:–URL identifies user

SIP URI or ‘tel’:–URL

identifies services hosted by ASs e.g. [email protected]

User Identities User Identities

Every IM CN subsystem subscriber has an IMPI (IM Private User Identity). The private identity is assigned by the home-network operator and is used for AAA purposes. This identity, a NAI (Network Access Identifier), is of the form user@realm.

The IMPI identifies the subscription, e.g. IM service capability, not the user, and is permanently allocated to a user during their subscription with the home network.

Every IM CN subsystem subscriber has one or more IMPU. The public user identity/identities are employed by any user to request communications to other users.

The public user identity takes the form of a SIP URL or the ‘tel:’-URL format and has to be registered either explicitly or implicitly before the identity can be used to srcinate IMS sessions.

With the introduction of standardized presence, messaging, conferencing and group service capabilities, identities are needed to identify services and groups that are hosted by ASs. Release 6 introduced the PSI (Public Service Identity). This takes the form of a SIP URL or it could be in ‘tel:’-URL format. This URL or ‘tel:’-URL is not related to a user but to a service.

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LTE Evolved Packet Core Network

IMPI IMPU

Home domain name Authentication key Algorithms

Sequence number checking .

. .

Other Applications ISIM

USIM

IC Card

USIM, UICC and ISIM USIM, UICC and ISIM

The USIM is an application that represents and identifies a user and their association with a home environment in the provision of cellular network services. The USIM contains functions and data needed to identify and authenticate users when cellular network services are accessed. USIM-specific information must be protected against unauthorized access or alteration.

The USIM is always implemented in a removable integrated circuit card called the UICC (Universal Integrated Circuit Card). Access to IMS services is possible using 3GPP Release 99 UICCs. The UICC is a physically secure device that can be inserted and removed from terminal equipment. It can contain one or more applications, one of which must be the USIM.

The UICC can host other applications as well as the USIM. Examples include electronic banking and credit card services.

Access to IMS services is achieved by using an ISIM (IMS Services Identity Module) application. The ISIM only provides the necessary security features for the IMS. It resides on a UICC. ISIM-specific information must be protected against unauthorized access or alteration.

The ISIM includes the IMPI, at least one IMPU, the home network domain name, support for sequence number checking in the context of the IMS domain, and an authentication key.

The same framework for algorithms as specified for the USIM applies for the ISIM.

LT3604/v4.0 © Wray Castle Limited A.19

Appendix

UE P-CSCF I-CSCF_Optionally HSS S-CSCF

used in roaming cases 1. SIP Register (IMPI, IP address, etc)

SIP Diameter

3. SIP Register (IMPI, IP address, etc)

Authentication Required 4a. Diameter MAR (IMPI) 4b. Diameter MAA (IMPI, RAND, AUTN, XRES, CK, IK)

MAR = Multimedia Auth Request MAA = Multimedia Auth Answer

5. 4xx Authentication Challenge (IMPI, RAND, AUTN, CK, IK) 6. 4xx Auth Challenge

(IMPI, RAND, AUTN) Calculate Response

7. SIP Register (IMPI, Authentication response)

8a. Diameter UAR (IMPI) 8b. Diameter UAA

(S-CSCF)

9. SIP Register (IMPI, Authentication response) Response Checked

SIP Registration and IMS Authentication SIP Registration and IMS Authentication

With a Default EPS Bearer in place and a P-CSCF discovered, a VoLTE-capable UE can register for service with the IMS by sending a SIP Register message.

In non-roaming scenarios, the P-CSCF can directly interrogate the HSS to determine the subscriber’s S-CSCF. In the roaming case the visited P-CSCF will have to pass the query to either an I-CSCF or an S-CSCF in the subscriber’s home network for action (this option is shown in the diagram).

The SIP Register message, containing details such as the P-CSCF address/name, the UE’s public user identity (IMPU), private user identity (IMPI) and IP address, is passed to the appropriate S-CSCF.

‘Appropriate’ in this sense means an S-CSCF that has access to the AS that control services to which the user has subscribed. If an individual S-CSCF does not support connectivity to all of the required ASs it will be necessary to register the UE with more than one S-CSCF.

If the S-CSCF determines that the subscriber requires authentication it sends a Diameter CX-Authentication-Vector-Request to the HSS containing the UE’s IMPI, the HSS generates a set of authentication vectors (RAND, XRES, AUTN, CK, IK) and passes them back to the S-CSCF. The S-CSCF forwards these vectors to the P-CSCF in a SIP Authentication Challenge message, from which the P-CSCF strips the XRES, CK and IK parameters before forwarding to the UE.

The UE uses the authentication key data stored on the ISIM to process the challenge (and generates its own versions of CK and IK) and returns an Authentication Response to the network in a new SIP Register message. The new Register message follows the same path as the first one and arrives at the S-CSCF, which checks the returned Authentication Response against the XRES supplied by the HSS. If the challenge response is correct the S-CSCF updates the HSS with the terminal’s changed status and the HSS responds with details of the UE’s service profile and of the AS platforms associated with the user’s subscribed services.

The S-CSCF provides the required registration information to the set of service control platforms indicated by the HSS, enabling the UE to be available for the user’s subscribed services. A SIP 200 OK confirmation message is relayed from the S-CSCF to the P-CSCF and onwards to the UE, at which point the UE is regarded as being ready and available for IMS services.

Further Reading: 3GPP TS 23.228:5.2, 33.203 (IMS Security)

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LTE Evolved Packet Core Network

Originating

IMS Mobile-Originated to IMS Mobile-Terminated Session IMS Mobile-Originated to IMS Mobile-Terminated Session

The standard signalling flow for the establishment of a session between two IMS-connected UEs, as would be used for a GSMA IR.92 to GSMA IR.92 call, is shown in the diagram.

This example shows UEs that may be connected to different IMS or IP-CAN networks and omits the I-CSCF (which is optionally employed in connections between IMS environments) to simplify the diagram.

Further Reading: 3GPP TS 23.228:5.6.2 (MO leg) and 5.7.2 (MT leg)

CS-MGW MGCF IM-MGW S-CSCF IMS

SCC AS MME

1. GTPv2-C PS to CS

Request 2. ISUP IAM (STN-SR)

4. H.248 MGCP

7. Invite (Re-invite to IM-MGW)

9. Reroute Media to

SR-VCC PS to CS handover allows a call srcinally established with a local access leg extending across the EPS to be handed over to a standard CS circuit across a legacy access network. To allow this to happen, the media session that srcinally flowed from the local UE via the PDN-GW to the remote user must be realigned so that the session data flows through the IM-MGW on its way to the legacy CS core network.

SR-VCC handover is initiated by the MME sending an SR-VCC PS to CS Request over the Sv interface to an MSC Server. The message includes the user’s IMSI and their associated STN-SR and C-MSISDN in addition to target RAN details such as target Cell ID (for handover to GERAN) or target RNC ID (for handover to UTRAN).

If the MSC Server hosting the Sv interface is not the same as the MSC/MSC Server that will ultimately be controlling the handed-over call then the Sv MSC Server initiates an Inter-MSC handover to transfer responsibility for the call to a suitable CS network node. The call-controlling MSC Server uses the information contained in the handover message to arrange for resource allocation in the target GERAN/

UTRAN and also prepares the target CS-MGW to handle the call. It then contacts the IMS by sending an ISUP IAM to the IMS MGCF.

The MGCF instructs the IM-MGW (referred to below as the incoming IM-MGW) to begin establishing the required connections and transposes the content of the IAM into a SIP Invite. The diagram shows the MGCF forwarding the SIP Invite directly to the S-CSCF that is controlling the local subscriber’s service; in reality, the MGCF may be required to perform a lookup on an ENUM (E.164 Number Translation) database to relate the supplied STN-SR to a valid SIP-URI. It may then be required to forward the SIP Invite to an I-CSCF, which will perform a lookup on the HSS to discover the appropriate S-CSCF. These steps have been omitted to make the diagram simpler to follow.

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LTE Evolved Packet Core Network

LRF connection; the GSMA IR.92 profile mandates that networks supporting its version of VoLTE handle Emergency Calls within the IMS unless local rules or regulatory requirements specify otherwise.

3GPP have developed an IMS Emergency Call architecture (described in 3GPP Release 9 TS 23.167) which introduces another type of Call Session Control Function, the E-CSCF, which will be located in the same network as the P-CSCF (e.g. it will be in the ‘local’ network even for roaming UEs).

TS 23.167 states that emergency call detection can be performed by either the UE or the P-CSCF. If the UE is able to perform this function it will decide whether to route the call setup request via the IMS or whether to fallback to a CS-capable cell to make the call. The UE will base its decision on whether the IMS declared itself to support IMS Emergency Calls during SIP Registration and whether CS Fallback or SR-VCC support was signalled by the EPS during initial Attach or the most recent TAU.

If an IMS Emergency Session is selected (or if the UE is unable to make the determination) a SIP Invite is forwarded to the P-CSCF, which forwards it to the local E-CSCF. The E-CSCF will contact the appropriate PSAP (Public Safety Answer Point, or emergency call centre) and will determine the appropriate route over which to create the media path; if the PSAP is reachable via an IP network the session may continue to use VoIP protocols; if not, the E-CSCF will invoke the MGCF/BGCF and route the session via an IM-MGW to the PSTN.

Further Reading: GSMA IR.92 Profile: Annex A, 3GPP TS 23.167

Appendix

Billing System

Operator's Network

MRCF

S-CSCF

MGCF

BGCF

PDN-GW CCF

Rf Rf Rf Rf Rf Rf Rf Rf Rf

I-CSCF

P-CSCF

PDN-GW

IMS Charging IMS Charging

Charging data is collected from a variety of IMS nodes, including the S-CSCF and P-CSCF, the MGCF and the BGCF.

In document NEGOCIACIONES DEL ALCA” (página 103-110)