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APAN 2003 Fukuoka
2003-01-23
Internet Application Technology Lab.Dept. of Computer Science, Chonnam National Univ.
Kugsang Jeong ([email protected])
The development of SIP based VoIP service
2SIP based VoIP
Contents
SIP based VoIP
SIP testbed
VoIP services
New service development CTM Reservation Call Service Simultaneous Inform Service
Call Measurement
conclusion
3SIP based VoIP
1. SIP based VoIP
VoIP Transmission of voice/video over IP-based data network Market driver
Cost saving Integration of data and voice to create new services and applications
Requirements Availability, scalability, voice quality
VoIP protocol
RTCPRTP
IP
MGCP
Call Control and SignalingSignaling and
Gateway Control
Media
H.225
Q.931
H.323
TCP
RAS
UDP
SIPH.245
Audio/Video
RTSP
4SIP based VoIP
1. SIP based VoIP
SIP Features for future VoIP Simplicity
Scalability
Modularity
Internet-enabled
SIP in Market Many products but no deployment case
lack of multimedia applications & services
Interoperability SIP – SIP, SIP – H.323, SIP – PSTN & Intelligent Network Service
5SIP based VoIP
1. SIP based VoIP
SIP Architecture
1
2
3
45
67
8
9
1011
12
SIP Client
SIP RedirectServer
SIP Proxy SIP Proxy
SIP Client(User AgentServer)
Location Service
Request
Response
6SIP based VoIP
2. SIP testbed
SIP Open Source: VOCAL Vovida Open Communication Library
An open source, IP centric communication software, development platform and library.
It runs on: Linux and Solaris operating systems. Intel (I86) based hardware.
provides
SIP Based Call Control and Switching
Operation System Support
Feature and Application Creation
7SIP based VoIP
2. SIP testbed
VOCAL architecture
CDR Server(s)
Feature Server(s)
Redirect Server(s)
Provisioning Server(s)
Policy Server(s)
Heartbeat Server
3rd Party Billing System
RADIUS
SNMP NetworkManager
ClearingHouse
Internet
Marshal ServerMarshal Server
PSTN
Gateway
Marshal Server
SIP IP Phone MGCP Device
MGCP/SIPTranslator
Marshal Server
H.323/SIP Translator
Marshal Server
H.323 Terminal
8SIP based VoIP
2. SIP testbed
Basic SIP Call using VOCAL
2.INVITE3.302
4.INVITE
8. 180 (RING)
9. 200 (OK)
10. ACK
Audio over RTP Channels
5. INVITE6.302
1.INVITE
Redirect Server
Marshal Server A
SIP PhoneUser BSIP Phone
User A
Marshal Server B
7. INVITE
9SIP based VoIP
2. SIP testbed
SIP based VoIP Testbed
redirect server / translator
marshal server marshal server
feature server
pintel Xpressa
Ubiquity 3.0Linphone 0.8
pintel Xpressa
Ubiquity 3.0Linphone 0.8
NetmeetingRay PhoneSIP <-> H.323SIP <-> H.323
• spec: P III 800, 512 RAM• OS: wow Linux 7.1 ( 2.4.2-3)
10SIP based VoIP
3. SIP based Services
Call Forwarding
Redirect Server
Marshal Server A
SIP PhoneUser A
Marshal Server C
SIP PhoneUser C
Feature Server
INVITE INVITE
ACK
INVITE
302
302
ACK302ACK
INVITE
INVITE
ACK
302
INVITE
302
ACK
INVITE
INVITE
SIP Messages:INVITE – User is invited to participate in session.ACK – Acknowledgement.302 – Moved temporarily.
11SIP based VoIP
3. SIP based Services
Call blocking
Redirect Server
Marshal Server
SIP Phone Marshal Server
Feature Server
INVITE INVITE
302
ACK
INVITE
403
ACK403
SIP Messages:INVITE – User is invited to participate in session.ACK – Acknowledgement.302 – Moved temporarily.403 – Forbidden.
Provisioning Server
SIPGateway
PSTN
call_blocking.cpl
User dials:
1-900-NNN-NNNN
12SIP based VoIP
3. SIP based Services
Service classification
Who registered services? How’s a call initiated?
Caller Callee CPBS (by caller) CTBS (by server)
Call blocking
Reservation call
Alarm call
Call screening
Call forward
Call forwarding
Call blocking
Call screening
Reservation call
Alarm call
•CPBS(Call Processing Based Service)
call is initiated by user and is processed as user’s demand
•CTBS(Call Time Based Service)
server initiated a call on reserved time.
13SIP based VoIP
3. SIP based Services
CPL Call Processing Language A Signaling protocol independent language Proposed by IETF XML-based CPL has been accepted as a proposed standard in IESG in Feb, 2002 Signaling server: Handles the routing issues of an internet phone call.
o CPL is appropriate for CPBSo So We propose new module for CTBS
on timecondition
true
false
truecall
CPL CTM
register register
call call
14SIP based VoIP
4. New service development
o common module for reservation based service
o generate SIP message on the reserved time
o used for reservation call, alarm call, etc.
CTM (Call Time Module)
o to establish a call when users want to calll
o to reserve call time & callee info.
Reservation Call Service
o to send users messages simultaneously on a certain time
o to register call time, user group info., messages
Simultaneous Inform Service
15SIP based VoIP
4. New service development
CTM operation
User SIP Component
rsv alarm etc
Reservation timeRegistered service
reservation
etc
alarm
Register service send SIP message
Generate SIP msg.
(INVITE)
Notify service with info.
Save info for service
DB SIP generator
16SIP based VoIP
4. New service development
CTM call control
ControllerSIP Phone
User A
INVITE
200 OK
ACK
INVITE
ACK
SIP PhoneUser B
200 OK
Audio over RTP Channels
CTMSIP Phone
User A
INVITE
200 OK
ACK
INVITE
SIP PhoneUser B
200 OK
Audio over RTP Channels
<basic flow> <CTM flow>
o Internet draft
o One entity (“controller”) sets up and manages a communication relationship between two others
3rd party call control
17SIP based VoIP
4. New service development
CTM functions
Reservation Agent
Service_Request()Service_save()
Call_Creator Agent
Rsv()Alarm()
Timer Agent
Polling()Time_Search()Type_Service
Call_Transfer Agent
Open_Socket()Send_Packet()Close_Socket()
Delete_Service()
SIP UA
Web or SIP UA
18SIP based VoIP
4. New service development
CTM display
• OS: Window, Linux• Language: Java (JDK 1.4), PHP• DB: MySQL
19SIP based VoIP
4. New service development
SIP UA for CTM
start
Receive SIP msg.
Retrieve callee info.From SIP header
Reservation call
New SIP msg generation
end
Session setup & termination
Ringing for rsv call Ringing for general call
no
20SIP based VoIP
4. New service development
Reservation Call Service using CTM
DB
UA A UA B
CTM
2
Save service
Register service
Info:time, Caller, Callee
1
Connection
10
4
5
Proxy Sever Redirect Sever
7
8
6
9
Send SIP msg
3
Polling to serve in time
INVITE
INVITE, 302
INVITE
200 OK
ACK
21SIP based VoIP
4. New service development
Reservation Call Service using CTM
SIP Message
Register serviceRinging for reservation call• OS: Linux• Language: C, GTK• Linphone SIP UA upgrade
22SIP based VoIP
4. New service development
Simultaneous Inform Service using CTM
UA A UA D
2
save service
polling to serve in time
Group,time info, inform msg.
1
3send SIP & inform msg.
UA B UA C
DB
23SIP based VoIP
4. New service development
Simultaneous Inform Service using CTM
Message
<Callee><Caller>ACK
• OS: Linux • Language:Java, GTK• Linphone upgrade
확인확인
24SIP based VoIP
5. Call Measurement
4. SIP Measurement
o gather measurement data from each clients
o user can check QoS of the call on web pages
Measurement Server
o measure data using ‘libpcap’o parameters: packet loss, delay, call time
o NTP is used for time sync.
Measurement Client
25SIP based VoIP
5. Call Measurement
4. VoIP Measurement (H.323 & SIP)- Configuration
LinphoneA
LinphoneB
RTP Connection
Measurement Server
DB
Caller_RTP_in
SIP session
callsignal
Caller_RTP_out Callee_RTP_in
Callee_RTP_out
gather measurement data
26SIP based VoIP
5. Call Measurement
4. SIP Measurement - Web Page
http://168.131.161.165/voip
27SIP based VoIP
6. Conclusion
SIP Good session protocol for voice/multimedia over IP
Service development CTM Reservation Call service Simultaneous Inform Service
SIP measurement in UA
Future plan CTM based Conferencing System 3rd party call controller
28SIP based VoIP
7. Reference sites
IETF SIP RFC document http://www.ietf.org/rfc/rfc3261.txt
IETF CPL RFC document http://www.ietf.org/rfc/rfc2824.txt
IETF 3pcc draft document http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-02.txt
Vovida Vocal system http://www.vovida.org
SIP center http://www.sipcenter.org
Internet2 VoIP WG http://netlab.indiana.edu/i2_voip_working_group/
AARNet VoIP http://www.aarnet.net.au/serivces/voip
APAN-KR VoIP WG http://voip.kr.apan.net