APAN 2003 Fukuoka

2003-01-23

Internet Application Technology Lab.Dept. of Computer Science, Chonnam National Univ.

Kugsang Jeong (handeum@iat.chonnam.ac.kr)

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