CINEMA Columbia InterNet Extensible Multimedia Architecture Kundan Singh [email protected]du Agenda • Basic IP telephony • Infrastructure and components • Additional services • Current and future work Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu and Min Yan “Internet Real Time Lab”
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CINEMA Columbia InterNet Extensible Multimedia Architecture Kundan Singh [email protected] Agenda Basic IP telephony Infrastructure and components.
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User information:• Contact location• Profile (e.g., password)• Aliases• Address book
System information• Configuration
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipd
Web based configuration
Web server
User profile can be modified using web browser. • Creating new user (admin/normal)• Changing profile and contact information (“follow me” service).• Web CGI scripts• Both sipd and web scripts use the database
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipd
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Web based configuration
Web server
IP phones connected to the departmental LAN. Users are identified by id, e.g., “[email protected]”
Software (sipc) for desktop. Allows audio, video, chat, white board, device control, instant message, presence and desktop sharing.
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipd
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
Phones128.59.19.233
Web based configuration
Web server
Phones register themselves with sipd when powered up.
• Constantly monitor the SIP server from remote• Report any problem (e.g., unauthorized registration request)• Use Simple Network Management Protocol (SNMP) –
client server• SIP Management Information Base (MIB) – describe
variables and and parameters for monitoring and control• SIP server implements MIB, can be controlled remotely by
an SNMP client.
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipdT1/E1 RTP/SIP
Telephone
SIP/PSTN Gateway
Department PBX Web based
configuration
Web server
SNMP agent for SIP MIB. Allows remote monitoring and control of the SIP server. (e.g., prompt when an unauthorized registration is attempted)
Telephoneswitch
SNMP(Network Management)
ServicesServicesInteroperability with H.323Interoperability with H.323
• H.323 – Proposed by ITU-T• Based on previous H.324/H.320 architecture for video
conferencing• Collection of other specifications (for signaling, control,
transport, etc.)• Defines everything, from supported codecs to controlling
remote camera. (vs SIP, define only what is needed)• NetMeeting – most commonly known client• http://www.packetizer.com - good introduction on H.323
(biased comparison with SIP!!)
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipdT1/E1 RTP/SIP
Telephone
SIP/PSTN Gateway
Department PBX Web based
configuration
Web server
Telephoneswitch
SNMP(Network Management)
SIPH.323convertor
NetMeetingsiph323
H.323
Allows an H323 client (Netmeeting) to use the services of our SIP infrastructure.
• Single messaging system – email, voicemail, fax/video, etc.• Voice mail and answering machine – first step• Streaming media for storage and retrieval – Real-Time
Streaming Protocol • RTSP – similar to HTTP, request-response, uses RTP• RealPlayer, QuickTime – popular clients• SIP for signaling of IP telephony calls• Access from telephone using a gateway
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipdT1/E1 RTP/SIP
Telephone
SIP/PSTN Gateway
Department PBX Web based
configuration
Web server
Telephoneswitch
SNMP(Network Management)
SIPH.323convertor
NetMeetingsiph323
H.323
Provides a voice mail and answering machine service to all the registered users. Has web interface for accessing voice mails.
rtspd
SIP/RTSPUnified
messaging
RTSP media server
sipum
Quicktime
RTSP clients
RTSPGeneric media server for playback and recording of messages. Can work with existing RTSP client, Apple’s QuickTime.
ServicesServicesConferencingConferencing
• Multiparty conferencing using SIP and RTP• Models – centralized, decentralized, multicast• Audio mixing, you don’t get back your own audio• Decode-add-encode• Video replication• Centralized SIP control• Centralized RTP mixer
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipdT1/E1 RTP/SIP
Telephone
SIP/PSTN Gateway
Department PBX Web based
configuration
Web server
Telephoneswitch
SNMP(Network Management)
SIPH.323convertor
NetMeetingsiph323
H.323
rtspd
SIP/RTSPUnified
messaging
RTSP media server
sipum
Quicktime
RTSP clients
RTSP
Centralized conferencing server for audio and video. Users can join from IP as well as PSTN.
SummarySummaryOur IP telephony test-bedOur IP telephony test-bed
• For implementing new services• For test and performance measurement• For replacing the plain old PBX
SummarySummaryMore InformationMore Information
More information on the system designhttp://www.cs.columbia.edu/~kns10/publication/cinematr.pdfhttp://www.cs.columbia.edu/~kns10/publication/cinematr.pdf
More information on the software components:http://www.cs.columbia.edu/~hgs/softwarehttp://www.cs.columbia.edu/~hgs/software
Questions asked during Questions asked during the talksthe talks
Some of the questions that (I think) were raised and answered by the speaker during the talk.1.Why not use IP address instead of SSRC in RTP?2.What are the authentication mechanisms used in SIP?3.Why is it not LGPL licensed? Are there other LGPL’ed SIP implementations? Does Columbia want to make money out of this?4.What is the capacity of the system? How many users, calls can it handle?5.Actual quality of audio, if we have tested between two IP phones on the real Internet? 6.Do you support any kind of encryption?
Additional slidesAdditional slides
ComponentsComponentsVoIP: Call from POTS to IPVoIP: Call from POTS to IP
1. Convert telephone number to a SIP address (user@domain)2. ENUM: Use DNS lookup
If you dialed 9397042Convert to +12129397042Do DNS lookup for 2.4.0.7.9.3.9.2.1.2.1.arpa.inResult will be a SIP URL, e.g., sip:[email protected]
IVRIVR1 platform platform• Voice and telephony functions (ASR2, TTS3, DTMF4)• Service logic (application specific)
• Receives incoming PSTN5 call• Responds back with prompts• Accepts user input (DTMF or speech)• Takes action based on user input
(Usually the service logic is programmed for the specific
application, say weather report)
[1] Interactive voice response[2] Automated speech recognition[3] Text to speech [4] Dual tone multi-frequency (touch tone)[5] Public switched telephone network
1-212-8545224
ComponentsComponentsIVR – on internetIVR – on internet
PSTN
End userEnd user
IVR platformIVR platform• Voice and telephony functions (ASR, TTS, DTMF)• Service logic (application specific)
End userEnd userVoice gatewayVoice gateway• Voice and telephony functions
Internet
Web serverWeb server
• Service logic
ComponentsComponentsIVR – on InternetIVR – on Internet
PSTN
End userEnd user
Internet
Voice gatewayVoice gateway
Web serverWeb server
• Service logic (CGI, servlet, JSP)
• Voice and telephony functions• VoiceXML browser
End userEnd userVXMLVXML HTMLHTML
DB
Multimedia
Audio/grammar
Scripts
Web server
ComponentsComponentsVoiceXML ExampleVoiceXML Example
<form><field name=“drink”> <prompt>Would you like Coffee, Tea, Milk or Nothing.</prompt> <option value=“coffee”>coffee</option> <option value=“tea”>tea</option> <option value=“milk”>milk</option> <option value=“nothing”>nothing</option></field><block> <submit next=“http://…/bartender.cgi” namelist=“drink”/></block></form>
ComponentsComponentsMiscellaneous Miscellaneous
1. Media Streaming2. Multiparty conferencing3. Voice mail and answering machine4. Unified messaging5. Event notification and presence6. Collaborative work environment