happy diwali a festival of light diwali or deepaawali means an array of lamps i.e.rows of diyas...
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Happy DiwaliA Festival of Light
Diwali or Deepaawali means an Array of Lamps i.e .Rows of diyas (Deep = Lamp, Vali =Array). Of all the festivals celebrated in India, Diwali is by far the most glamorous and important. Enthusiastically enjoyed by people of every religion, its magical and radiant touch creates an atmosphere of joy and festivity. As a family festival, it is celebrated 20 days after Dussehra, on the 13th day of the dark fortnight of the month of Ashwin (October / November). This year it falls on 14th Nov. It is a festival of lights symbolizing the victory of righteousness and the lifting of spiritual darkness. It celebrates the victory of good over evil - and the glory of light. This festival commemorates Lord Rama's return to his kingdom Ayodhya after completing his 14-year exile. Homes are decorated, sweets are distributed by everyone and thousands of lamps lit to create a world of fantasy. Diwali is a time for fun and revelry. Diwali is also a time for pooja and tradition.
An Overview of CINEMA Implementation
OverviewOverview
• Modules
• Applications
• Performance
• Misc – Compilation
– Installation
– Portability
• Current and Future work
Presented by: Kundan Singh
Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran
Narayanan, Henning Schulzrinne and Min Yan
Nov 14, 2001IRT Group Meeting
3
OverviewMultimedia Communication Protocols
Physical layer
Link layer
Network (IPv4, IPv6)
Transport (TCP, UDP)
Application layer
H.323 RTSP RSVP RTCP
RTP
MediaG.711MPEG
SIP
SignalingQuality of service
Media transport
InternetTelephony
InternetRadio/TV Messaging
and Presence
Interactivevoice response
Unified messagingVideo
conferencing
4
OverviewCINEMA - Columbia InterNet Extensible Multimedia
Architecture
CINEMA Libraries
CINEMA Applications
A flexible architecture to support wide range of multimedia A flexible architecture to support wide range of multimedia communication applications, both clients and serverscommunication applications, both clients and servers
Proxy server, media server, voice mail, conferencing, etc.
Parsing, SIP, SDP, RTP, mySQL interface, SNMP interface, Portability stubs, etc.
5
OverviewSIP and sipd
• Address based on email ([email protected])
Columbia.edu
Cisco.com home.com
office.com
Alice
Bob
1. DNS home.compc1.home.compc1.home.com 129.59.19.140
2. INVITE [email protected]. INVITE [email protected]
(proxy mode)
(2)(3) m2.home.com
sipd
6
Overviewsipd – Example scenario
Bob(1)
(2)
(3)(4)
(5)
(6) (6)
(6)
(7)
Alice
(8)(9)
(10)
(11)
(12)
(13)
7
An Overview of CINEMA Implementation
OverviewOverview
• ModulesModules
• Applications
• Performance
• Misc – Compilation
– Installation
– Portability
• Current and Future work
Various CINEMA libraries and their functionality
8
ModulesFunctionality
• Message parsing: SIP, RTSP• Transaction state and client branch• User agent call state• Interface to external modules: database, SNMP• Higher level policy: sip-cgi• Canonicalize: e.g., Henning.Schulzrinne => hgs• Authentication: basic, digest
9
ModulesMessage Parsing (libcine)
HTTP
RTSP SIP
GET /sip HTTP/1.0Host: www.cs.columbia.edu…
DESCRIBE rtsp://… RTSP/1.0Accept: application/sdp… INVITE sip:… SIP/2.0
From: kns10@cs…
• Utilities for URL, headers, constructing and parsing messages
10
ModulesTransaction state (libsip)
• A request and all its responses
• RTSP vs SIP requests• Request can be
• Proxied• Redirected• Generated• Terminated
11
ModulesCall state (libsip++)
12
Modulescanonicalization [libcanon]
Bob.Wilson
canonicalize
bob@cs
13
ModulesLibraries
sipd sip323 sipconf sipum sipvxmlrtspd
CINEMA Libraries
libNT
Win32 stub
libcine
Utilities parsingIPv6
libsip
Basic SIP library
libsip++
SIP UA library
libmixer
RTP audio mixer
libdict
Hash table
libdb++
mySQL intf
RTSP mediaserver
SIP proxy server
SIP/H.323gateway
SIP/RTP conferencing
SIP/RTSP unified messaging
SIP/VoiceXMLbrowser
LDAPXerces-C
OpenH323
MySQLPWLibResparse
librtsp
RTSPclient
librtp
RTP library
libsnmp
SIP MIB
ViaVoiceXerces-C
CINEMA Applications
Parsing, SIP, SDP, RTP, mySQL interface, SNMP interface, Portability stubs, etc.
14
ModulesLayered structure
Transport layer (TCP/UDP)
RTPInterface
HTTP Message Parsing
RTSP transaction
SIP transactionClient Branch
RTSP API
RTSP server
SIPUA API
SIP proxy
Other Applications
15
An Overview of CINEMA Implementation
OverviewOverview
• Modules
• ApplicationsApplications
• Performance
• Misc – Compilation
– Installation
– Portability
• Current and Future work
Our test-bed architecture and its components
16
Test-bed ArchitectureTest-bed Architecture
SIP proxy,redirectserver
SQLdatabase
sipd
Proxy, Redirect, Registration server.• Authentication• Programmable (SIP- CGI)
OpenSource SQL database: MySQL
http://www.mysql.com
User information:• Contact location• Profile (e.g., password)• Aliases• Address book
System information• Configuration
17
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
18
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.
19
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.
Sipd stores the contact information in the database table:[email protected] => [email protected]
There can be multiple contacts. All registered phones ring, and the first to pick up is connected.
Registration can also be altered from the web interface
20
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
(2) Phone rings, the user picks up the call and can talk to the caller.
(1) When somebody calls [email protected], sipd gets the INVITE message and “proxies” the call to the current location.
Another IP phone
Based on user profile, sipd may ask for caller authentication.
21
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
Cisco 2600 router with SIP/PSTN gateway connects the departmental LAN with the PBX.
Departmental PBX (Nortel Meridian) connects both internal and external lines to the gateway.
Telephoneswitch
Internal T1
External T1
(Extension:713x)
Dial “8” to reach outside line
22
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
(2) The gateway forwards all PSTN calls to sipd; sip:[email protected]
(1) When PBX receives a call for 9397132, it forwards the call to extension 7132. 7130-7139 is assigned to the gateway.
Telephoneswitch
Internal T1
(Extension:713x)
Dials 9397132
128.59.19.141
(3) Sipd looks into the dialplan, finds a mapping 7132=>[email protected] and forwards the call to the current location of “hgs”.
23
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 Gateway128.59.19.28
Department PBX Web based
configuration
Web server(2) Sipd authenticates the caller
and checks permissions. Sipd maps the number 5551212 to [email protected], adding the prefix “8” and the gateway address.
(3) PBX forwards the call to external line.
Telephoneswitch
128.59.19.141
(1) The IP user dials “sip:[email protected]”.
Dial “8” to reach outside lineExternal T1
(4) PSTN user receives the call.
24
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
Can use no-DID (direct inward dialing) mode for more numbers.
Telephone number mappings and privileges modifiable from the web
Telephoneswitch
Internal T1
External T1
25
Additional ServicesAdditional Services
• Advantage: cost savings + new services• “Think of receiving your voicemail messages in an email
that you can later play out in a conference to show it to others”
• Easy integration of email, web, instant messaging, etc.• Open architecture vs Close architecture (traditional
telephones)
26
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
Programmable server: SIP-CGI and Call Processing Language (CPL)
Telephoneswitch
Scripts can be uploaded by clients also.
27
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)
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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.
29
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.
30
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.
SIP conference
server
sipconf
31
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
Netmeeting user dials “sip:[email protected]”
SIP conference
server
sipconf
SIP user dials “sip:[email protected]”
PSTN user dials 1-212-9397139
Sipd maps 7139=> [email protected]
128.59.19.196
32
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Our IP telephony test-bedOur IP telephony test-bed
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
SIP conference
server
sipconf
Device GW
X 10
W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards Junking the PBX: Deploying IP Telephony". NOSSDAV 2001,
33
PSTN to IP CallPSTN to IP Call
PBXPSTN
External T1/CAS
Regular phone(internal)
Call 93971341
SIP server
sipd
Ethernet
3
SQLdatabase
4 7134 => bob
sipc
5
Bob’s phone
• Direct Inward Dial (DID) - direct and simple• No-DID - dial extension, supports more users
GatewayInternal T1/CAS(Ext:7130-7139)
Call 71342
713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX
34
IP to PSTN CallIP to PSTN Call
Gateway(10.0.2.3)
3
SQLdatabase
2Use sip:[email protected]
Ethernet
SIP server
sipdsipc
1Bob calls 5551212
PSTN
External T1/CASCall 55512125
5551212
PBX
Internal T1/CASCall 85551212 4
Regular phone(internal, 7054)
Note: In this direction there is no distinction between DID and non-DID calls.
35
An Overview of CINEMA Implementation
OverviewOverview
• Modules
• Applications
• PerformancePerformance
• Misc – Compilation
– Installation
– Portability
• Current and Future work
Discussion of some performance issues and solutions
36
PerformanceWhy is it important ?
• Reduce server cost per user => more users per server
• Registration: 100 requests/s => 180000 registered users (1 hr refresh time, digest authentication)
• Call: 100 requests/s => 120000 calls per hour (record route)
• Bandwidth: 1 Gb/s => (approx) 6250 bi-directional G.711 simultaneous participants in a conference.
37
PerformanceFor which components ?
• Signaling: proxy, registrar (sipd)– Receive message
– Act on it (canonicalize, database lookup)
– Proxy the message
– Send/proxy response back
• Media: sipconf, rtspd– Some processing for each media stream
– File I/O
– Encode/decode (audio mixing)
– Forward packets (video)
38
PerformanceThreads
• One thread per request– For 1MB virtual memory per thread on 32 bit machine: max
limit of 4000 threads. OS limits (for regular user 1024 pthreads on solaris)
– 30 second wait per stateful INVITE request; limits to 130 R/s
– Thread creation overhead
1. Customize stack size
2. Use thread pool or event model
39
PerformanceThread pool
• Request are put in an event queue
• Worker threads pick up the event and execute
• Fix the number of worker threads
• G/G/T/N queue
• Use thread pool for all requests
• Need to rewrite sleep/wakeup so that 30 sec wait does not waste a thread
• Will multiple process help? may be for stateless proxies
40
PerformanceIn-memory DB
• Every query to database affects turn-around time
• Duplicate the DB in main memory; hash-table
• Less than 4k per user (?)
• Replacement algorithm? Not needed
• Synchronization: separate threads– Primary user table, aliases: relatively static, readonly by sipd,
refresh every 30 min
– Contacts table: read-write, refreshed every 2 min
– Read only modified records since last read, write back only modified records
41
PerformanceDatabase
• NFS issues (log, scripts, database files?)
• SQL logging: currently serialized; use lazy write back; logging at the end of request processing, so it does not affect response time but consumes resources (worker thread) for longer time per request
• DB on same machine or on remote machine?
42
PerformanceBandwidth
• Assuming avg message length 130 bytes; on 100 Mb/s with effective 40%, 3000 requests/s
• Affects more to media components– Number of simultaneous media streams served by rtspd
– Number of participants in a conference by sipconf
– Number of simultaneous three party conferences by sipconf
43
PerformanceGeneral Comments
• Measure performance on various platforms (Linux, Solaris Netra, Dec Alpha)
• Compare stateless vs stateful proxy
• Compare in-memory (fastsql) vs database (sql)
•
•
•
44
An Overview of CINEMA Implementation
OverviewOverview
• Modules
• Applications
• Performance
• MiscMisc – Compilation
– Installation
– Portability
• Current and Future work
45
Compilation
• Autoconf, configure and make for Unix platforms– Solaris, Linux, FreeBSD, Tru64
$ ./configure –with-mysql=… --with-…
$ make sipconf
• Microsoft VC++ 6.0 for Windows NT/2000
• Makefile.in (global), module.mk (per module)
46
InstallationAnd software distribution
• GUI based configuration
• Package manager (Sun, Linux, FreeBSD,…), Installation scripts, windows install shield
• Monitor scripts, RC
47
PortabilityCross platform support
• Endian-ness: Big endian (Sparc, DEC), little endian (Intel)
• 32 bit vs 64 bit
• Unix vs Windows standard libraries (threads)
• Re-entrant APIs (gethostbyname_r, strtok_r)
• NTutils for win32
• Shared libraries compilation (?)
48
Documentation
• software documentation: – http://www.cs.columbia.edu/IRT/cinema
• Overview:– Paper: http://www.cs.columbia.edu/~hgs/papers/Jian0106_Junking.pdf
– Tech report (incomplete): http://www.cs.columbia.edu/~kns10/publication/cinematr.pdf
• Compilation instructions:– README, README.build, NT/README.win32 files
• This presentation:– http://www.cs.columbia.edu/~kns10/talks (will be put up shortly)
49
Current and Future Work
• Improved installation
• Address book
• Calendar and event notification
• Conference recording (local file, media server)
• File sharing in a conference from web
• Voice dialogs for conferencing and voicemail
• Load balancing on multiple conference servers
• Conference (floor) control from web
50
Current and Future Work
• From a multimedia communication test bed to a multimedia collaboration portal environment
• Scaling to large call volumes and users
51
PublicationsFor more information
• W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards Junking the PBX: Deploying IP Telephony". NOSSDAV 2001,
• H. Schulzrinne, S. Narayanan, J. Lennox and M. Doyle, “SIPstone – Benchmarking SIP Server Performance”. Aug 2001. http://www.sipstone.org
• Kundan Singh, Gautam Nair and Henning Schulzrinne, "Centralized Conferencing using SIP". Proceedings of the 2nd IP-Telephony Workshop (IPTel'2001), April 2001.
• K. Singh, H.Schulzrinne, "Interworking Between SIP/SDP and H.323". Proceedings of the 1st IP-Telephony Workshop (IPTel'2000), April 2000.
• Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP". IP Telecom Services Workshop 2000, Sept 2000. Atlanta, Georgia, U.S.A.