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LOGO 1 Mobile Internet Telephony: Mobility Extension to H.323 Adviser: Ho-Ting Wu Speaker: Chih-Hao...
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Transcript of LOGO 1 Mobile Internet Telephony: Mobility Extension to H.323 Adviser: Ho-Ting Wu Speaker: Chih-Hao...
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LOGO
Mobile Internet Telephony: Mobility Extension to H.323
Adviser: Ho-Ting Wu
Speaker: Chih-Hao Tseng
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OutlineIntroduction VoIPMobility Issues On H.323 TermainalsCall Signaling Procedure For Mobility Managem
entConclusionReference
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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.
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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)
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RTP & RTCP Sequencing Payload Identification Frame Indication Source Identification Synchronization Transport Address
QoS feedback Session control Identification Synchronization
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VoIP PhonesProtocol
SJphone SIP & H.323 (Optional)
Skype Independent
Gizmo SIP
JAJAH SIP
OpenH323 H.323
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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
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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
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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
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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
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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
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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
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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)
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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: [email protected] Sip: [email protected]
Supporting Protocol Session Description Protocol (SDP) Session Announcement Protocol (SAP)
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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
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Invitation
Location(Redirect) server
Proxy
tsengii(1) INVITE
tsengii
key
(2)
tse
ng
ii
(3)
tsen
gii
@14
0.12
4.18
1.20
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(4) INVITE [email protected]
(5) 200 OK(6) 200 OK
(7) ACK [email protected] (8) ACK [email protected]
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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
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Mobility Issues On H.323 Terminals
Internet Telephony: From stationary to mobile Terminals
System ArchitectureMobility Management
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abbreviation
POTS: Plain Old Telephone ServiceIAM: Initial Address MessageACM: Address Complete MessageANM: ANswer MessageREL: RELease
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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
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Handoff mechanism
Home agent Home location register
Foreign agent Visitor location register
Real-time handshaking
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System Architecture
Subnet 5
Subnet 4
Subnet 3
Client
Subnet 2Subnet 1
ClientClient
Roaming
Roaming
Client
Client
ClientClient
Client
Roaming
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Intrazone roaming
A mobile host moves with a zone
Subnet 2Subnet 1
Client Client
Gatekeeper
roaming
ITG
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Interzone roaming
It crosses the boundary to other zones.
Subnet 2Subnet 1
ClientClient
Gatekeeper
Roaming
ITG
Gatekeeper
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Call Signaling Procedure For Mobility Management
RegistrationCall EstablishmentRoamingUsing IP Multicast to Support Mobility
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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.
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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
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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.
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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.
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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.
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Conclusion(2/2)Signal exchange in ITG
Circuit switching to Packet switching
Handoff time delay Too long to make session disconnected?
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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