Transcript Mobile VOIP

VoIP & Mobile VoIP
97598039 梁紀翔
NETLab
Topics
► Voice




over Internet Protocol
H.323, SIP, Skype
Adoption
Benefits
Challenge
► Mobile
VoIP
 Generic Access Network
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Voice over Internet Protocol
► Family
of transmission technologies for
delivery of voice communications over IP
networks
► Talks to other IP phone or PTSN
► Use codecs convert analog voice into digital
audio stream
► Video, text, file, conference, voice mail…etc
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Common protocols
► H.323
(ITU-T)
► Session Initiation Protocol (RFC 3261)
► Skype (Propritary)
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H.323
► First
VOIP standard (Feb. 1996)
► Part of H.32x family
► Audio-Video communication on packet
network
► Call singling & control, multimedia transport
& control, bandwidth control for P2P and
multipoint conference
► Based on Q.931 of ISDN
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Audio Coding/Decoding
(G.711, etc)
Management
H.225
Call
Control
Signaling
H.225
RAS
(Registration,
Admission,
and Status)
TCP
System conrtrol
H.245
Terminal
Control
&
RTP/RTCP
(Real-time Transport
protocol/Real-time
Transport Control
Protocol)
UDP
IP
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Gatekeeper Multipoint Control Units
Gatekeeper Multipoint Control Units
LAN
LAN
WAN
Terminal
Terminal
Gateway
Gateway
Zone
Zone
PSTN
ISDN
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.
1.1
H.
R
5
2
2
AS
P
D
(U
)
1.2
.H
.22
Gatekeeper
5R
AS
( UD
P)
2.H.225 Call Setup (TCP)
3.H.245 Call Control (TCP)
4.RTP (UDP)
Caller
Callee
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SIP
► Application
Layer protocol for controlling
multimedia communication session over IP
► Client-Server HTTP like Text-based control
message request/response
► Depend on SDP (RFC 2327), RTP
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Request
Messages Type
Description
Response
Messages Type
Description
REGISTER
Registers the user agent
1xx
Informational Responses
ex. 180 Ringing
INVITE
Initiates a call by inviting
user to participate in
session
2xx
Successful Responses
ex. 200 OK
3xx
ACK
Confirms that the client has
received a final
response to an INVITE
request
Redirection Responses
ex. 302 Moved
Temporarily
4xx
Request Failure Responses
ex. 404 Not Found
5xx
Server Failure Responses
ex. 503 Service
Unavailable
6xx
Global Failures Responses
ex. 600 Busy Everywhere
CANCEL
Cancels a pending request
BYE
Indicates termination of the
call
OPTIONS
Used to query the
capabilities of a server
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Skype
► Proprietary
P2P model,
written by KaZaA
developers
► Decentralized user directory
► AES encrypted
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Adoptions
► Many
implementations of
H.323 & SIP
 Analog Telephone Adapter
connects old phone and IP
network
 Dedicated IP phone
 Softphone on PC
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Benefits
► Reducing
costs
communication and infrastructure
 Charged per amount of data not per time
 Free/Open-source IP-PBX software integrate
legacy phone system ex. Asterisk
► Flexibility
 Easy to encrypt, location independence
 Integration with other services
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Challenge
► QoS
 Latency and Jitter
► Power
failure
► Emergency call
► Lack of redundancy
► Number portability
► Security
► Fax and other telephony devices
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Mobile VoIP
► Extension
of mobility to VoIP network
► Mobile device with SIP client ex. fring
 High speed IP communication 3.5G, WiFi,
WiMax
 Some company may block SIP
► Use
softswitch to bridge SIP and RTP to cell
phone network
 Handset still use GSM or CDMA
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Generic Access Network
► 3GPP
standard to let dual-mode handset to
seamlessly roam or handover between
WLAN and WAN(GPRS)
► GAN Controller
 Inside cellular core network, as another base
station
► When
detects WLAN, establish secure IP
connection to GAN Controller
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Questions ?
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References
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VoIP http://en.wikipedia.org/wiki/VoIP
H.323 http://hive.packetizer.com/users/packetizer/papers/h323/h323_protocol_overview.pdf
SIP http://www.cs.columbia.edu/sip/
Asterik http://www.asterisk.org/
Skype http://www.skype.com/intl/en/
GAN http://www.3gpp.org/ftp/Specs/html-info/43318.htm
GAN http://www.3gpp.org/ftp/Specs/html-info/44318.htm
“行動網際網路電話之研製與實現”, 李昆達, 2005
“H.323 網路電話視訊監控與錄製系統之研製”, 曾致皓, 2008
“H.323 網路電話音訊監控與錄製系統之研製”, 黃威穎, 2008
“客製化H.323協定之智慧型網路電話監控與錄音系統”, 蔡家瑞, 2009
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