The development of SIP based VoIP service

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The development of SIP based VoIP service
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Contents

• SIP based VoIP
• SIP testbed
• VoIP services
• New service development
• CTM
• Reservation Call Service
• Simultaneous Inform Service
• Call Measurement
• conclusion

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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

Media
Signaling and Gateway Control
Call Control and Signaling
Audio/ Video
H.323
H.225
RTCP
RTP
MGCP
Q.931
RAS
SIP
H.245
RTSP
TCP
UDP
IP
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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

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1. SIP based VoIP

• SIP Architecture

Request
SIP Redirect Server
Response
Location Service
2
3
5
4
6
1
7
11
10
12
SIP Proxy
SIP Proxy
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9
SIP Client
SIP Client (User Agent Server)
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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

Operation System Support
Feature and Application Creation
SIP Based Call Control and Switching
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2. SIP testbed

• VOCAL architecture

MGCP/SIP Translator
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2. SIP testbed

• Basic SIP Call using VOCAL

2.INVITE 3.302
5. INVITE 6.302
4.INVITE
8. 180 (RING)
7. INVITE
1.INVITE
9. 200 (OK)
Audio over RTP Channels
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2. SIP testbed

• SIP based VoIP Testbed

redirect server / translator
SIP lt-gt H.323
Ray Phone
marshal server
marshal server
pintel Xpressa
pintel Xpressa
feature server

• spec P III 800, 512 RAM
• OS wow Linux 7.1 ( 2.4.2-3)

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3. SIP based Services

• Call Forwarding

SIP Messages INVITE User is invited to
participate in session. ACK Acknowledgement. 302
Moved temporarily.
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3. SIP based Services

• Call blocking

call_blocking.cpl
User dials 1-900-NNN-NNNN
SIP Messages INVITE User is invited to
participate in session. ACK Acknowledgement. 302
Moved temporarily. 403 Forbidden.
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3. SIP based Services

• Service classification

• CPBS(Call Processing Based Service)
• call is initiated by user and is processed as
  users demand
• CTBS(Call Time Based Service)
• server initiated a call on reserved time.

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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.
• CPL is appropriate for CPBS
• So We propose new module for CTBS

CPL
CTM
register
register
condition
true
on time
true
call
call
call
false
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4. New service development
CTM (Call Time Module)

• common module for reservation based service
• generate SIP message on the reserved time
• used for reservation call, alarm call, etc.

Reservation Call Service

• to establish a call when users want to calll
• to reserve call time callee info.

Simultaneous Inform Service

• to send users messages simultaneously on a
  certain time
• to register call time, user group info., messages

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4. New service development

• CTM operation

DB
SIP generator
Reservation time Registered service
Notify service with info.
reservation
Generate SIP msg. (INVITE)
alarm
rsv
alarm
etc
Save info for service
etc
send SIP message
Register service
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4. New service development

• CTM call control

3rd party call control

• Internet draft
• One entity (controller) sets up and manages a
  communication relationship between two others

Audio over RTP Channels
ltbasic flowgt
ltCTM flowgt
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4. New service development

• CTM functions

Web or SIP UA
SIP UA
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4. New service development

• CTM display

• OS Window, Linux
• Language Java (JDK 1.4), PHP
• DB MySQL

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4. New service development

• SIP UA for CTM

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4. New service development

• Reservation Call Service using CTM

DB
2
Save service
3
Polling to serve in time
1
Register service Infotime, Caller, Callee
CTM
Redirect Sever
4
Proxy Sever
Send SIP msg
INVITE, 302
6
5
7
INVITE
INVITE
9
ACK
8
200 OK
Connection
10
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4. New service development

• Reservation Call Service using CTM

SIP Message
Register service
Ringing for reservation call

• OS Linux
• Language C, GTK
• Linphone SIP UA upgrade

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4. New service development

• Simultaneous Inform Service using CTM

save service polling to serve in time
2
DB
1
Group,time info, inform msg.
3
send SIP inform msg.
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4. New service development

• Simultaneous Inform Service using CTM

• OS Linux
• LanguageJava, GTK
• Linphone upgrade

Message
ACK
ltCalleegt
ltCallergt
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5. Call Measurement
4. SIP Measurement
Measurement Server

• gather measurement data from each clients
• user can check QoS of the call on web pages

Measurement Client

• measure data using libpcap
• parameters packet loss, delay, call time
• NTP is used for time sync.

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5. Call Measurement
4. VoIP Measurement (H.323 SIP) -
Configuration
callsignal
SIP session
RTP Connection
Caller_RTP_out
Callee_RTP_in
Caller_RTP_in
Callee_RTP_out
gather measurement data
DB
Measurement Server
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5. Call Measurement
4. SIP Measurement - Web Page
http//168.131.161.165/voip
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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

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7. Reference sites

• IETF SIP RFC document http//www.ietf.org/rfc/rfc3
  261.txt
• IETF CPL RFC document http//www.ietf.org/rfc/rfc2
  824.txt
• IETF 3pcc draft document http//www.ietf.org/inte
  rnet-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_voi
  p_working_group/
• AARNet VoIP http//www.aarnet.net.au/serivces/voip
  
• APAN-KR VoIP WG http//voip.kr.apan.net