1

LOGO

Mobile Internet Telephony: Mobility Extension to H.323

Adviser: Ho-Ting Wu

Speaker: Chih-Hao Tseng

2

OutlineIntroduction VoIPMobility Issues On H.323 TermainalsCall Signaling Procedure For Mobility Managem

entConclusionReference

3

IntroductionVoice over IP (VOIP) uses the Internet Protocol (IP)

to transmit voice as packets over an IP networkDigital signal processors (DSP) segment the voice

signal into frames and store them in voice packets. Here the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network.

Internet telephony gateway (ITG) bridges switched circuit phone networks and packet-switched data network.

4

Structure

PSTN

Intranet

ITG

Voice port

Router

Voice port

Router

5

VoIP

Signaling User Information Translation Session Establishment Session Negotiation

• agree on a set of media and codecs, add/suppres media, change encoding,…

Media Transport Real-time Transport Protocol (RTP) Real-time Transport Control Protocol (RTCP)

6

RTP & RTCP Sequencing Payload Identification Frame Indication Source Identification Synchronization Transport Address

QoS feedback Session control Identification Synchronization

7

VoIP PhonesProtocol

SJphone SIP & H.323 (Optional)

Skype Independent

Gizmo SIP

JAJAH SIP

OpenH323 H.323

8

Codec

ITU-T Standard Compression Method Bit Rate (kbps)

Complexity Coding Delay (ms)

G.711 PCM 64 1 0.75

G.726 ADPCM 32 10 1

G.728 LD-CELP 16 50 3 to 5

G.729 CS-ACELP 8 30 10

G.729a CS-ACELP 8 15 10

G.723.1 MP-MLQ 6.3 25 30

G.723.2 ACELP 5.3 25 30

9

Introduction H.323ITU-T standard

International Telecommunication Union-Telecommunication Standardization Sector

Provides the technical requirements for voice communication over IP service

Control Protocol H.225/Q.931 Call Signaling H.225 (Registration, Admission, Status)RAS Signaling H.245 Media Control

10

Components of H.323 Terminal

Endpoint that support H.245, Q.931, RAS, RTP,MCU

Gateway Connect heterogeneous networks

Gatekeeper Address Translation Bandwidth Management

Call signaling address

Call signaling

portRAS port

Endpoint type

E.164 address

H.323 ID TTL

11

RAS Message (1/2)Process Message acronym

Gatekeeper(GK) Searching

Gatekeeper Request GRQ

Gatekeeper Confirm GCF

Gatekeeper Reject GRJ

Endpoint Registrar

Registrar Request RRQ

Registrar Confirm RCF

Registrar Reject RRJ

Unregistered Request URQ

Unregistered Confirm URJ

Unregistered Reject UCF

Endpoint Located

Location Request LRQ

Location confirm LCF

Location Reject LRJ

12

RAS Message (2/2)

Admission Control

Admission Request ARQ

Admission Confirm ACF

Admission Reject ARJ

Disengage Calling

Disengage Request DRQ

Disengage Confirm DCF

Disengage Reject DRJ

Bandwidth management

Bandwidth Request BRQ

Bandwidth Confirm BCF

Bandwidth Reject BRJ

Status Checking

Information request IRQ

Information request response

IRR

Resource Indicator

Resource Availability Indicator

RAI

Resource Availability Confirm

RAC

13

Protocol Stack

Audio apps

Video apps

Terminal control and management

Data apps

G.711

G.722

G.723

G.728

G.729

H.261

H.263RTCP H.225

RAS signaling channel

H.225 Call

signaling channel

H.245 Control Channel

T.124

RTP X.224 Class T.125

Unreliable transport (UDP) Reliable transport

(TCP)

T.123

Network layer (IP)

Subnet layer

14

H.323 Call Setup Flow Chart

3. Setup

Q.931 Call Signaling Channel

PictureTel

RAS Channel

1. ARQ (alias address/bandwidth)

2. ACF (call signaling channel address/bandwidth)

5.ARQ

6.ACF

8. Connect (H.245 Address)

4. Call Proceeding7. Alerting

H.245 Control Channel (Logical Channel 0)

Master/Slave Determination

RTCP StreamRTCP Stream

RTP Stream

Gatekeeper

Terminal

PictureTel

Terminal

Capability Exchange

OpenLogicChannel (RTCP address)OpenLogicChannelACK (RTP & RTCP address)

15

Introduction SIP Session Initiation Protocol (SIP) IETF standard

Internet Engineering Task Force Application layer control protocol for creating, modifying and ter

minating sessions Text-encoded protocol

SIP URL Sip: bob@ntut.edu.tw Sip: 11025919@ntut.edu.tw

Supporting Protocol Session Description Protocol (SDP) Session Announcement Protocol (SAP)

16

Communication Flow Chart

17

Components of SIP Phone SIP Clients Proxy Server

acts as both a server and a client for the purpose of making requests on behalf of other clients

forwards client requests to another SIP server or to the final destination

may “fork” requests to multiple servers (“search tree”)

Registrar Server A server that receives registrations from clients and regarding

current locations

Redirect (Location) Server redirects users to try other SIP server as the next-hop towards

destination, and client has to contact it directly

18

Invitation

key@140.124.181.217

Location(Redirect) server

Proxy

tsengii@140.124.181.206

tsengii(1) INVITE

tsengii

key

(2)

tse

ng

ii

(3)

tsen

gii

@14

0.12

4.18

1.20

6

(4) INVITE tsengii@140.124.181.206

(5) 200 OK(6) 200 OK

(7) ACK tsengii@140.124.181.206 (8) ACK tsengii@140.124.181.206

19

SIP Message body

20

Compare between H.323 & SIP

Standardization sector

Server Degree of difficulty

Header field Cost Integrated

SIP software phone

IETF Proxy Server Easy to Implement

Texual representation

Low Incomplete

H.323 software phone

ITU-T Gatekeeper Hard to Implement

Binary representation

high More Complete

21

Supporting Protocols

22

Mobility Issues On H.323 Terminals

Internet Telephony: From stationary to mobile Terminals

System ArchitectureMobility Management

23

abbreviation

POTS: Plain Old Telephone ServiceIAM: Initial Address MessageACM: Address Complete MessageANM: ANswer MessageREL: RELease

24

POTS phone to H.323 terminal

25

H.323 terminal to POTS phone

26

Internet Telephony: From stationary to mobile Terminals

Static IP to Dynamic IPDiscrete reachability

Portability Off –line reachability

Continuous reachability Mobility (encompasses portability) On-going connection

27

Handoff mechanism

Home agent Home location register

Foreign agent Visitor location register

Real-time handshaking

28

System Architecture

Subnet 5

Subnet 4

Subnet 3

Client

Subnet 2Subnet 1

ClientClient

Roaming

Roaming

Client

Client

ClientClient

Client

Roaming

29

Mobility Management4 core operations:

Registration Call establishment Roaming Handoff

30

Intrazone roaming

A mobile host moves with a zone

Subnet 2Subnet 1

Client Client

Gatekeeper

roaming

ITG

31

Interzone roaming

It crosses the boundary to other zones.

Subnet 2Subnet 1

ClientClient

Gatekeeper

Roaming

ITG

Gatekeeper

32

Call Signaling Procedure For Mobility Management

RegistrationCall EstablishmentRoamingUsing IP Multicast to Support Mobility

33

RegistrationA Gatekeeper may advertise its availability for

mobility services (GK active) Gatekeeper multicasts a new message called

Gatekeeper advertisement (GAD) to the respective zone to advertise its availability.

A mobile terminal may solicit the service of a Gatekeeper (terminal active) A mobile terminal may send a GRQ message to

the Gatekeeper well-known discovery multicast address.

34

Call EstablishmentA call between a Plain Old Telephone Service

(POTS) phone and an H.323 terminal can be made from either direction. POTS phone to an H.323 cross ITG An H.323 to a POTS phone cross ITG

35

Call establishment ITG to H.323 terminal

36

Call establishment H.323 terminal to ITG

37

RoamingRequest-to-join

Requesting by the new participants Only the request-to-join approach is

demonstrated. Invite-to-join

Being invited by the participants of a conference Invite-to-join can be performed in a similar

manner.

38

Call signaling for roaming: intrazone roaming

39

Call signaling for roaming: interzone roaming

40

Using IP Multicast to Support Mobility

D class： 224.0.0.0～ 239.255.255.255： for Multicast

Individual hosts are free to join and leave a multicast group at any time

Sends an Internet Group Management Protocol (IGMP) leave message to the immediately neighboring multicast router to depart the group in the old subnet, and sends a report message to join the group in the new subnet.

41

Call signaling for call establishment with multicasting ITG to H.323

42

Call signaling for call establishment with multicasting H.323 to ITG

43

Call signaling for intrazone roaming with multicasting

44

Call signaling for interzone roaming with multicasting

45

Conclusion(1/2)Two approaches to mobility:

Using ad hoc multipoint conference expansions Using IP multicasting to emulate mobility

Since ad hoc multipoint expansion has been defined in H.323, our solution introduces no additional entities to H.323 and requires minimal modifications to the existing H.323 protocol.

46

Conclusion(2/2)Signal exchange in ITG

Circuit switching to Packet switching

Handoff time delay Too long to make session disconnected?

47

Reference

“Mobile Internet Telephony : Mobility Extension to H.323”, Wanjiun Liao, Member, IEEE,IEEE Transactions on vehicular technology, vol.50, no.6, November 2001.

OpenH323,” http://www.voxgratia.org/” OpenH323, “http://www.openh323.org/” http://life.iiietc.ncu.edu.tw/xms/forum/show.php?id=4125 校園網路 IPv6 SIP VoIP 之建置與推廣 , 黃悅民 , 國立成功大學 工程科學系

教授 http://www.environmental-studies.de/GSM_Netz/Roaming/R-3/Handover.jpg