ip technology, part 2

IP technology, part 2

1) Mobility aspects (terminal vs. personal mobility)

2) GPRS (IP traffic in mobile network)

3) Session Initiation Protocol (SIP)

4) Mobile IP (adding mobility to wireless network)

5) Quality of Service (QoS) support

6) ATM basics

Mobility in IP networks

Personal mobility (e.g. offered by SIP)

Terminal mobility (e.g. offered by GPRS)

The Mobile IP concept tries to combine both, when implemented together with wireless LAN technology.

One can very generally define two types of mobility:

The IMS (IP Multimedia Subsystem) concept in 3GPP Release 5 also tries to combine both (using SIP and GPRS technology).

User mobility vs. terminal mobility

Personal mobility (e.g. offered by SIP):

User can move around in the network and use a new terminal after registration via the new terminal. The new terminal has the same address for incoming calls as the old terminal.

Terminal mobility (e.g. offered by GPRS):

User can move around in the network and use the terminal at different locations => location updating. The user cannot change to a different terminal (without transferring the SIM to the new terminal).

GPRS (General Packet Radio Service)

GSM BSS

3G RAN

PS core network

CS core network

GGSNGGSNSGSNSGSN

HLRHLR

AuCAuC

EIREIR

PSTN

Inte

rnet

MS

Handles packet (IP) traffic in a mobile network

SGSN

GSM BSS

3G RAN

PS core network

CS core network

GGSNGGSN

HLRHLR

AuCAuC

EIREIR

PSTN

Inte

rnet

MS

SGSNSGSN

The serving GPRS support node in the packet-switched core network is the equivalent of the serving MSC/VLR in the CS core network.

GGSN

GSM BSS

3G RAN

PS core network

CS core network

SGSNSGSN

HLRHLR

AuCAuC

EIREIR

PSTN

Inte

rnet

MS

The gateway GPRS support node is the equivalent of the GMSC in the CS core network. IP traffic from external IP networks (e.g. the Internet) is always first routed to the GGSN.

GGSNGGSN

GPRS attach / PDP session

GPRS attach

MS is assigned PDP (IP) addressPacket transmission can take place

Separate or combined GSM/GPRS attachMS registers with an SGSN (authentication...)Location updates are now possible

PDP context is created

GPRS detach

PDP context is terminatedAllocated IP address is released

In case of dynamic address

allocation

DHCP(Dynamic Host Configuration

Protocol)

PDP context

Packet data protocol context describes characteristics of GPRS session (session = "always on" connection)

PDP context information is stored in MS, SGSN and GGSN

MSMS

GGSNGGSNSGSNSGSN

::::::::::::

::::::::::::

::::::::::::

PDP type (e.g. IPv4)

PDP address = IP address of MS (e.g. 123.12.223.9)

Requested QoS (priority, delay …)

Access Point Name = IP address of GGSN (e.g. 123.12.223.0)

PDP type (e.g. IPv4)

PDP address = IP address of MS (e.g. 123.12.223.9)

Requested QoS (priority, delay …)

Access Point Name = IP address of GGSN (e.g. 123.12.223.0)

One user may have several PDP context sessions active 123.12.223.9

123.12.223.0

PDP context activation

MSMS GGSNGGSNSGSNSGSN

::::::::::::

Activate PDP context request

Create PDP context request

Create PDP context response

Activate PDP context accept ::::::::::::

::::::::::::

IP address allocated to MS

Security functions

Packet transmission (1)

MS (client)

MS (client)

GGSNGGSN

SGSNSGSN Server Server

Dynamic IP address allocation has one problem:It is difficult to handle a mobile terminated transaction(external source does not know IP address of MS)

Fortunately, packet services are usually of client-server type => MS (not server) initiates packet transmission

GGSN knows temporary IP adress of user

GGSN knows temporary IP adress of user

Server has to find correct GGSN

Server has to find correct GGSN


MS (client)

MS (client)

GGSNGGSN

SGSNSGSN

Packet is sent to SGSN. SGSN sends packet to GGSN through GTP (GPRS Tunneling Protocol) tunnel.

Packet is tunneled through IP backbone of PS core network

Outer IP header Outer IP header IP headerIP header IP payloadIP payload

Tunneling = encapsulation of IP packet in GTP packet

Server Server


MS (client)

MS (client)

GGSNGGSN

SGSNSGSN

Outer IP header Outer IP header IP headerIP header IP payloadIP payload

Dest. IP address: GGSN

Dest. IP address: Server

Server Server

Note the different destination IP addresses


MS (client)

MS (client)

GGSNGGSN

SGSNSGSN

GGSN sends packet through the Internet to the server.

IP headerIP header IP payloadIP payload

Source IP address: GGSN, not MS

Dest. IP address: Server

Server Server


MS (client)

MS (client)

GGSNGGSN

SGSNSGSN

Server sends return packet via GGSN to MS. Note the different destination IP addresses along the way.

Packets from server to MS are always routed via GGSN (since this node has PDP context information, including the dynamically allocated IP address of MS).

Dest. IP address:

GGSN

Dest. IP address: MS

Server Server

Dest. IP address (in outer IP header): SGSN

Session Initiation Protocol (SIP)

SIP is a protocol for establishing, controlling and releasing anything from simple voice calls (VoIP) to complex multiparty multimedia sessions over IP networks (i.e., packet switched networks).

SIP was designed by the IETF (Internet Engineering Task Force). As a result, SIP closely resembles the HyperText Transfer Protocol (HTTP) which takes care of client - server type of transactions in the Internet.

Contrary to HTTP, SIP can handle client – client (or peer-to-peer) type of communications through IP networks.

Separation of control and user plane

The routing in the control plane (the SIP signalling) and user plane (transport of the multimedia data) is separate.

User B User B

User C User C

User A User A

SIP signalling

Multimedia data

http://www.ietf.org/rfc/rfc3261.txt

SIP vs. H.323

The ITU-T based H.323 standard describes a suite of protocols for managing multiparty multimedia calls using circuit switched technology.

Microsoft’s NetMeeting is based on H.323. Since the operation of H.323 is rather complex, it is outside the scope of this course.

IETF-based SIP is less complex and can perform most of the call/session control tasks of H.323.

Call control in the IMS (IP Multimedia Subsystem) as defined by 3GPP (3rd Generation Partnership Project) is based on SIP.

H.323 protocol suite

ApplicationApplication

CodecsCodecs

RTPRTP

Terminal / application controlTerminal / application control

RTCPRTCP H.225.0RAS

H.225.0RAS

H.225.0CS

H.225.0CS H.245H.245

UDPUDP TCPTCP

Network layer (IP)Network layer (IP)

Data link layerData link layer

Physical layerPhysical layer

User plane Control plane (signalling)

(You do not have to remember this slide)

H.323 protocol suite (control part) (You do not have to remember this slide)


CodecsCodecs

RTPRTP

Terminal / application controlTerminal / application control

UDPUDP TCPTCP

Network layer (IP)Network layer (IP)

Data link layerData link layer

Physical layerPhysical layer

User plane Control plane (signalling)

Registration, Admission and Status

Call control Signalling

Media stream management between endpoints

RTCPRTCP H.225.0RAS

H.225.0RAS

H.225.0CS

H.225.0CS H.245H.245

RTP Control Protocol

SIP architecture

Location Service

SIP Registrar

SIP Proxy

SIP User Agent (Calling user)

SIP User Agent (Called user)

SIP Proxy

SIP operation (registration)

Location Service

SIP Registrar


SIP Proxy

This means binding the current IP address (e.g. the IP address in the 802.11 subnetwork or the PDP address in GPRS) with the globally known SIP address. From this point onwards the SIP user can be reached.

Before a SIP user can be reached, (s)he must first register. Address

Binding

SIP address

Location ServiceGlobally unique SIP address is point

of contact from the outside world:

sip:[email protected]

After registration, a location server (in SIP terms: Location Service) will contain a binding between SIP address and registered IP address.

sip:[email protected]

124.2.10.57

124.2.10.57

SIP operation (basic call 1)

Location Service

SIP Proxy



SIP Proxy

Another SIP user "calls this SIP address". The call is routed to the registered IP address.

1

2

3

4

124.2.10.57


SIP Proxy



SIP Proxy

During the call setup, the called user is also given the IP address of the calling user (since both ends must know the other end’s IP address).

124.2.10.5798.13.0.44

This is my address!




After completed call setup, the end users communicate in a normal fashion over the IP network(s).

The SIP architectural entities shown on the previous slides are not involved in this user plane communication.

124.2.10.5798.13.0.44 IP network(s)

Voice over RTP over UDP over IP (e.g.)

SIP offers the following features

Call forking (several terminals at various locations are alerted at the same time)

Capability of multimedia calls (codecs for voice, video) can be negotiated using Session Description Protocol (SDP) messages carried over SIP

User-friendly addressing (sip:[email protected])

Personal mobility (but not terminal mobility)

Good flexibility, scalability, extensibility

Interworking between SIP telephony and PSTN telephony (as well as between SIP addressing and E.164 addressing).

Basic (two-party) SIP call (1)

User A User A

SIP proxy of user A

User B User B

"Invite" message (corresponding to IAM message in ISUP) is sent to SIP proxy of user A. The message includes SIP address (sip:[email protected]) of user B.

SIP address of Alice:sip:[email protected]

Invite ...

Request

Response

[email protected]

BobIP address of Alice’s current terminal


User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

"Invite" message is routed to SIP proxy of user B (Bob). How does SIP proxy of Bob know where Bob is at this moment? (At home, at work, ...?)

Invite ...

[email protected] [email protected]

BobIP address of Alice

SIP registration

User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

The answer is: The terminal of Bob has performed SIP registration. After registration, the SIP proxy of user B knows the IP address of Bob’s current terminal.

Register ...

[email protected]

IP address of Bob

[email protected]

IP address of Alice

IP address of Bob’s current

terminal


User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

"Invite" message is routed to Bob’s terminal using the IP address provided via SIP registration. Alice’s IP address is included in the message.

Invite [email protected]@net1.com

IP address of Alice IP address of Bob


User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

Bob’s terminal is ringing. An (optional) "180 Ringing" message is routed back to user A (Alice) and an audio ringing tone is generated in Alice’s terminal.

180 Ringing

[email protected]@net1.com



User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

Bob answers the call. A "200 ok" message is routed back to Alice. Alice sends an "Ack" message to Bob (not necessarily via the SIP proxies, since Alice’s terminal now knows the IP address of Bob’s terminal).

200 ok

Ack

[email protected]@net1.com



User A User A

SIP proxy of user A

User B User B

SIP proxy of user B

The user plane data (e.g. VoIP over RTP) is transported between the terminals without involving SIP proxies.


SIP forking example

User A User A

SIP proxy of user A

Terminal 1 Terminal 1

SIP proxy of user B

Forking: different terminals of user B are alerted at the same time. The one that answers first returns the "200 ok" message ...



Terminals of user B that have performed

SIP registration

Invite ...

Three types of addresses

E.164 address Address points directly to called user in the PSTN358 9 1234567

MSISDN Address points to HLR in home network of called user040 1234567

SIP address Address points to SIP proxy of called usersip:[email protected]

HLR knows to which serving MSC/VLR the call should be routed

SIP proxy knows where to route "Invite" SIP message

What can SIP do?

The most important task of SIP is to find out URLs of terminals to be included in the multimedia session (see example).

For negotiation of multimedia capabilities, SIP can carry SDP messages between end users (in "Invite" and "200 ok" SIP messages).

Unfortunately, SIP cannot influence the transport in the user plane (support of QoS and security features, inclusion of PCM/EFR transcoding equipment, etc.).

Mobile IP

Another way of implementing mobility in an IP network is to use Mobile IP.

Like SIP, Mobile IP requires some specialised nodes in the IP network (Home Agent and Foreign Agent) for handling mobility. In fact, all mobility schemes require some point of contact where information about the current location of the called party is stored:

Mobility concept Point of contact

2G/3G mobile networks Home Location Register (HLR)SIP Location Service (server)Mobile IP Home Address (Home Agent)


Basic Mobile IP architecture

Mobile Node (roaming terminal)

Correspondent Node (terminal in external

network)

Home address (public IP address of mobile user) points to user database

in Home Agent

Care-of Address (Temporary IP

address in visited network)

Visited network

Operation: 1. Discovering Care-of Address

Mobile Node Public address

Home AddressCare-of Address

Mobile Node obtains Care-of Address.

Rfc 2002 defines certain procedures how this can be done.

Proprietary solutions in 802.11 (WLAN) networks?

124.2.10.57 54.12.9.11

Operation: 2. Registering Care-of Address

Mobile Node

Registration is necessary, so that incoming traffic (i.e. packets intended for the mobile node) can be routed to the Care-of Address.


Public address

54.12.9.11124.2.10.57

124.2.10.57

Route incoming traffic to this

address

Operation: 3. Incoming traffic

Mobile Node

Correspondent Node

Incoming traffic is first routed to Home Address and is then tunneled to Care-of Address (known by the Home Agent).


54.12.9.11124.2.10.57

98.13.0.44

Operation: 4. Tunneling

Mobile Node

Tunneling means packing the original IP packet into the payload of an outer IP packet (IP over IP).


54.12.9.11124.2.10.57

H Payload

H Payload Original IP packet

Outer IP packet

Operation: 5. Outgoing traffic

Mobile NodeOutgoing traffic need not be routed via Home Agent (mobile node knows IP address of correspondent node).


54.12.9.11124.2.10.57

Correspondent Node

98.13.0.44

Note that source IP address in outgoing packet is Home Address, not Care-of Address

QoS support in IP networks

"Best effort" service <=> no Quality of Service support

Some alternatives for introducing QoS in IP backbone applications:

Alternative 1: RSVP (Resource ReSerVation Protocol)

Alternative 2: DiffServ (Differentiated Services)

Alternative 3: MPLS (MultiProtocol Label Switching)

Alternative 4: IP tunneling over ATM

IETF terminology: Traffic engineering

Problems with “Best effort” IP transport

"Best effort" service is sufficient for traditional Internet applications like web browsing, e-mail, and file transfer.

"Best effort" is not sufficient for real-time applications:

Speech (voice)Speech (voice)

Multimedia applicationsMultimedia applications

Video / audio streaming

Video / audio streaming

Low round-trip delayLow round-trip delay

Low delay Low delay

Low delay variation Low delay variation

Consistent throughputConsistent throughput

High throughputHigh throughput

QoS support mechanisms (1)

RSVP (Resource ReSerVation Protocol)

RSVP can be considered an example of the integrated services concept (compare with differentiated services).RSVP is typically used together with other mechanism(s).

IP Backbone

Ingress point

Egress point

Resources are reserved beforehand (or at certain intervals)

HostHost HostHost



DiffServ (Differentiated Services)

IP Backbone

Ingress point

Egress point

Service "tagging" in ToS byte at ingress point

HostHost HostHost

Traffic control based on ToS byteIPv4 Header

VersionVersion IHLIHL Type of ServiceType of Service Total lengthTotal length

IdentificationIdentification FlagsFlags

Time-to-liveTime-to-live ProtocolProtocol HeaderHeader

ToS byte = 8 bits (28 = 256 priority levels

could be used, but only few are used)



MPLS (Multi-Protocol Label Switching)

Virtual connection must be established first (using e.g. RSVP).

IP datagrams are encapsulated in MPLS frames and relayed through the backbone (=> MPLS label is used for routing).

MPLS Backbone

Ingress point

Egress point

Label switching in all routers along the path

HostHost HostHost

LSR = Label Switch Router (router with MPLS functionality)


QoS support mechanisms (3 cont.)

MPLS label structure:

L2 payloadL2 payload L2 HL2 H

IP datagram Header of layer 2 protocol data unit

L2 payloadL2 payload LabelLabel L2 HL2 H

TTL (8 bits)TTL (8 bits) SS ExpExp Label value (20 bits)Label value (20 bits)

Label length = 32 bits

L2 PayloadL2 Payload LabelLabel LabelLabel LabelLabel L2 HL2 H

Stack bit identifies bottom-of-stack label

Stacking:

Label at top of stack is always in use first

routing without MPLS

in case of MPLS

QoS support mechanisms (3 cont.)

Routing without MPLS: destination IP address in IP header is used for routing.

L2 payload L2 payload LabelLabel L2 HL2 H

L2 payload L2 payload L2 HL2 HDA

DA

In case of MPLS: destination IP address is not used for routing along the virtual path between ingress and egress point. Routing is based on MPLS label instead.


IP tunneling over ATM

ATM Backbone

Ingress point

Egress point

IP packets are directed to the ingress point

HostHost HostHost

IP traffic is carried over ATM virtual connection

More about ATM on the next slides.

see also: http://www.infosyssec.com/infosyssec/secatm1.htm

ATM is a transport technique


IPIP

ATMATM

PhysicalPhysical


IPIP

PhysicalPhysical

Network backbone based on ATM

Network backbone based on TDM

Network backbone based on IP


PhysicalPhysical


ATMATM

PhysicalPhysical

PoS (Packet over SDH/Sonet)Application can be circuit

switched or packet switched (possibility of IP-over-IP)

IPoA

for implementing network "backbone"

Typical ATM network connection

Higher layersHigher layersHigher layersHigher layers

AALAAL AALAAL

ATM layerATM layer ATM layerATM layer ATM layerATM layer

PhyPhy PhyPhy PhyPhy

Originating node ATM network nodes

User to Network Interface (UNI)

ATM layerATM layer

ATM switching within the ATM layer

PhyPhy PhyPhy PhyPhy

Terminating node

User to Network Interface (UNI)

Network to Network Interface (NNI)

ATM protocol reference model

Segmentation and reassembly (SAR),flow & error control

ATM cell multiplexing and switching

Usually SDH transport (STM-N)

ATM adaptation layer

ATM layer

Physical layer

User application (higher protocol layers)

E.g., IP traffic

Functions of the physical layer

The physical medium sublayer (lower sublayer) handles and adapts bits as they are fed to the physical medium (e.g., electro-optical conversion)

The transmission convergence sublayer (upper sublayer) converts the flow of cells from the ATM layer into a continuous bit stream (and vice versa), involving:

• Cell rate decoupling (bit rate adaptation)

• Cell delineation (generally using HEC method)

• Mapping of cells into, e.g., SDH VC-4 payload

• Calculation and verification of HEC byte

SOH

SOH

STM-1 payload

VC-4 (Virtual container)

VC-4 (Virtual container) POH

AU-4 pointer points to first byte of VC

3

5

9 261 bytes

...

...

1 260 bytes

VC "floats" in STM-1 frames

ATM cell

Mapping of ATM cells into STM-1 frames

Cell delineation (1)

Cell delineation = finding the borders between cells at the receiving end of an ATM link

... ...received bit stream ... where does a cell begin?

Method 1: using VC-4 POH pointer

POH ...

VC-4 (Virtual container)ATM cell

Cell delineation (2)

Method 2 is usually preferred over Method 1 (better performance)

Method 2: using the HEC byte

We take 4 x 8 consecutive bits from the received bit stream and calculate the checksum

If the checksum = the next byte (= HEC byte), we have found the header of an ATM cell. If not, we shift one bit position and repeat the calculation ...

Checksum = HEC byte ?

Functions of the ATM layer

ATM cell creation => generating and adding a 5 byte cell header to the 48 byte payload received from the AAL (and giving the payload to the AAL at the receiving end)

Multiplexing (and demultiplexing) of the ATM cell flows from different sources using appropriate identifiers (VCI and VPI) located in the cell headers

Cell routing (switching and/or relaying) within the ATM network (also using VCI and VPI)

The ATM layer may also provide mechanisms for traffic management.

1.

2.

3.

4.

Bits

Byte in header

ATM cell header structure

8 7 6 5 4 3 2 1

GFC* / VPI VPI

VPI VCI

VCI

VCI

PTI

HEC

GFC Generic Flow Control HEC Header Error ControlVPI Virtual Path Identifier PTI Payload Type IndicatorVCI Virtual Channel Identifier CLP Cell Loss Priority

* GFC at User to Network Interface (UNI)

CLP

VPI and VCI

8 7 6 5 4 3 2 1

GFC* / VPI VPI

VPI VCI

VCI

VCI

PTI

HEC

CLP

VPI and VCI are virtual connection identifiers (used for switching or relaying the cells to the correct destination)

Physical layer

connection(e.g. STM-1)

VPI 1

VPI 2

VCI 1

VCI 2

Multiplexing in AAL 2

Multiplexing in ATM layer

One-bit Cell Loss Priority (CLP)

8 7 6 5 4 3 2 1

GFC* / VPI VPI

VPI VCI

VCI

VCI

PTI

HEC

CLP

Cell Loss Priority bit is used to indicate the priority of the cell

In case of network congestion cells with lower priority will be discarded first.

Three-bit Payload Type Indicator (PTI)

8 7 6 5 4 3 2 1

GFC* / VPI VPI

VPI VCI

VCI

VCI

PTI

HEC

CLP

One PTI bit is used in AAL 5

User data / control data

Traffic management

0 0 0 User data cell, no congestion. ATM-user indication = 00 0 1 User data cell, no congestion. ATM-user indication = 10 1 0 User data cell, congestion. ATM-user indication = 00 1 1 User data cell, congestion. ATM-user indication = 11 0 0 OAM F5 segment associated cell1 0 1 OAM F5 end-to-end associated cell1 1 0 Resource management cell1 1 1 Reserved for future VC functions

Header Error Control (HEC)

8 7 6 5 4 3 2 1

GFC* / VPI VPI

VPI VCI

VCI

VCI

PTI

HEC

CLP

Only bit errors in ATM cell header are checked, not bit errors in cell payload

At the transmitting side, the checksum is calculated over the four first header bytes. The result is inserted into the HEC field.

At the receiving side, the HEC byte may be used for error control and cell delineation purposes (as shown on a previous slide).

ATM Adaptation Layer (AAL)

1. Flow & error control, handling of lost and misinserted cells (not widely used…)

2. Segmentation and reassembly (SAR) of data to fit into ATM cells as cell payload.

ATM cell payloadATM cell payload HdrHdr

Data block Hdr SAR PDU

ATM Layer PDU (= ATM cell)

Implemented in the end-point nodes only (routing is not addressed, this is covered by the ATM layer).

Higher layer data (e.g. IP datagram)

AAL protocols

For real-time data that requires constant bit rate, small delay, small delay variation (PCM speech transport)

Like AAL 1, but offers multiplexing of data from different sources into a single ATM cell. Used mainly for compressed voice & video

For data that is not time sensitive (e.g. IP transport); simple SAR, no retransmission mechanisms (see next slide).

AAL 1

AAL 2

AAL 5

Example: AAL 5 operation

HH

SAR-PDUSAR-PDU

CS-PDU info (< 65532 bytes)

User info (e.g. IP datagram)

SAR-PDUSAR-PDUSAR-PDUSAR-PDU

PayloadPayload

SAR-PDUs

Last SAR-PDU => PTI LSB = 1 ATM cell

PTI LSB = 0

8 bytes: length of info field, CRC error checking

pad

N x 48 bytes

(PTI is in ATM cell header)

48 bytes

ip technology, part 2

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