MPLS - Presentation Transcript
MPLS ( Multiprotocol Label Switching ) Oleh : Sunaryo Tandi N I M : 0801050005
Pengertian MPLS
Multiprotocol Label Switching (disingkat menjadi MPLS ) adalah teknologi penyampaian paket pada jaringan backbone berkecepatan tinggi.
Asas kerjanya menggabungkan beberapa kelebihan dari sistem komunikasi circuit-switched dan packet-switched yang melahirkan teknologi yang lebih baik dari keduanya
MPLS berada di antara lapisan kedua dan ketiga.
OSI
OSI sendiri merupakan singkatan dari Open System Interconnection . Model ini disebut juga dengan model " Model tujuh lapis OSI " ( OSI seven layer model ).
Managed IP Services Layer 4+ Data Services Contoh Aplikasi per Layer OSI System Integrator Layer 7 Application & Outsourcing Service Provider Layer 2 - 3 IP VPN Wireless Data Metro Optical Ethernet Internet Access Data Transport TDM Transport Enhanced Communication Transaction Processing Application Hosting Application Development System Management System Integration Business Process Consulting Hardware Management &support Network Operation & Management Application Management Content Delivery LAN Management IP Voice PBX/IP Centrex Storage Disaster Recovery Web Hosting Managed Security Video Conferencing
Prinsip kerja MPLS Routing Qos POLICY Forwading PAKET OUT PAKET IN SIGNALLING
Prinsip kerja MPLS
Menggabungkan kecepatan switching pada layer 2 dengan kemampuan routing dan skalabilitas pada layer 3
menyelipkan label di antara header layer 2 dan layer 3 pada paket yang diteruskan
Label dihasilkan oleh Label-Switching Router (LSR)
Label berisi informasi tujuan node selanjutnya kemana paket harus dikirim
Paket-paket diteruskan dalam path yang disebut LSP ( Label Switching Path ).
MPLS di Hirarki Network
Komponen MPLS
Label Switched Path (LSP): Merupakan jalur yang melalui satu atau serangkaian LSR dimana paket diteruskan oleh label swapping dari satu MPLS node ke MPLS node yang lain.
Label Switching Router : MPLS node yang mampu meneruskan paket-paket layer-3
MPLS Edge Node atau Label Edge Router (LER) : MPLS node yang menghubungkan sebuah MPLS domain dengan node yang berada diluar MPLS domain
MPLS Egress Node : MPLS node yang mengatur trafik saat meninggalkan MPLS domain
MPLS ingress Node : MPLS node yang mengatur trafik saat akan memasuki MPLS domain
MPLS label : merupakan label yang ditempatkan sebagai MPLS header
MPLS node : node yang menjalankan MPLS. MPLS node ini sebagai control protokol yang akan meneruskan paket berdasarkan label.
Keuntungan Menggunakan MPLS
MPLS benefits include better performance, lower total cost of ownership, greater flexibility to accommodate new technologies, better security and survivability.
Better performance: Uses Classes of Service (CoS/QoS) and priority queuing so your network knows which traffic is most important and ensures that it takes priority over other traffic.
Depending on your current enterprise class network, you can reduce your on-going WAN operating costs by up to 50%, while maintaining a high level of reliability and service.
“ Future-proof” the architecture of your network so it can respond rapidly to changing business needs (e.g. New services, latency sensitive traffic, bandwidth intensive traffic , VoIP, video).
Lower packet loss means faster response for many applications.
Keuntungan Menggunakan MPLS
Network survivability from its fully meshed nature.
Consolidate your network to a single, enterprise-wide view of your sites/group of companies.
Have the option to deliver firewalled internet access from the cloud to specified facilities to eliminate internet local loop costs
Reduce the time and cost involved in managing a technologically disparate “system of systems”.
Online reporting allows you to truly see what is happening on your network so you subscribe only to the bandwidth that you really need.
Simplify the administration and on-going management of your network.
VPN dengan MPLS
Salah satu feature MPLS adalah kemampuan membentuk tunnel atau virtual circuit yang melintasi networknya. Kemampuan ini membuat MPLS berfungsi sebagai platform alami untuk membangun virtual private network (VPN).
VPN yang dibangun dengan MPLS sangat berbeda dengan VPN yang hanya dibangun berdasarkan teknologi IP
VPN pada MPLS lebih mirip dengan virtual circuit dari FR atau ATM, yang dibangun dengan membentuk isolasi trafik.
Lapisan pengamanan tambahan seperti IPSec dapat diaplikasikan untuk data security.
Keuntungan VPN MPLS
Paket data dikirimkan berdasarkan kode-kode yang ada pada label. Tiap paket data yang dikirim akan membawa sebuah label yang mengindentifikasikan tujuannya.
Memungkinkan untuk membuat konfigurasi mesh dalam jasa penyelenggara telekomunikasi, tidak perlu dikonfigurasikan sendiri oleh pelanggan (Jaringan cost-effective fully-mesh topologies )
Tidak membutuhkan perangkat tambahan (seperti halnya IP Sec via Internet) di sisi pelanggan – enskapsulation MPLS terjadi di dalam jaringan penyelenggara
Memungkinkan bundling value added services ke dalam MPLS-VPN (Internet, voice dan data secara bersamaan).
Terima Kasih
Source http://www.slideshare.net/idnats/mpls
title : Lewat FastFlip, Google Jadikan Internet Bagai Majalah
summary : Google memperkenalkan cara baru mereka dalam menampilkan berita-berita dari internet. Berbagai situs web ditampilkan bagai majalah. (read more)
If you have ever used a real remote computer system like Citrix, then you have probably been craving multiple Remote Desktop sessions since you first fired up Windows XP Professional and/or Media Center Edition. Here is a HACK (translated: USE AT YOUR OWN RISK), to enable multiple Remote Desktop sessions on your XP Pro or MCE 2005 box:
NOTE: You will have to have knowledge of the Windows operating system and more specifically the Windows Registry. If you have no experience with the registry, then I would recommend you find someone who does or leave these alone. I do not make any kind of warranty that this will work for you or your friends. This is provided for entertainment purposes only. Don’t call me if your computer stops working. Got it?
- Print these directions so that you have them to work from.
- Restart your computer in Safe Mode - Follow this link to learn how to restart Windows XP in Safe Mode
- Turn off/disable Remote Desktop Connection (RDC) and Terminal Services
- Right click My Computer
- Select Properties
- Click on the Remote tab at the top of the window
- UNCHECK the box next to, “Allow users to connect remotely to this computer“
- Click OK
- Go to Start -> Control Panel -> Administrative Tools -> Services
- Find Terminal Services in the list
- Right click on Terminal Services and click Properties
- In the Startup Type box, select Disabled
- Click OK to close the window
- Here is a copy of the Terminal Services DLL - Save it to your Desktop or other suitable location
- Using a file manager like Windows Explorer open C:\Windows\system32\dllcache
- Rename the file termsrv.dll to termsrv_dll.bak or whatever you would like.
- Copy the downloaded termsrv.dll file (the one you just downloaded from the web) to C:\Windows\system32\dllcache
- Open the C:\Windows\system32 folder
- Delete the file termsrv.dll in C:\Windows\system32
You should be good to go.
Sering kali kita merasa disulitkan oleh banyaknya tanda pagar (#) pada file-file di linux, padahal yang akan belum tentu kita merasa perlu untuk membaca tulisan yang ada di belakang tanda tersebut.
Berikut cara simple untuk melihat suatu file konfigurasi pada linux tanpa menampilkan tanda pagar (#). Lewat command line interface, jalankan perintah berikut;
cat /lokasi/file/file.conf | sed '/ *#/d; /^ *$/d'
Open Source VOIP applications, both clients and servers.
Open source means all source code is available!! Do not post any "free but not open" software here!
SIP Proxies
Mini-SIP-Proxy A very tiny perl POE based SIP proxy
MjServer: cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
MySIPSwitch: SIP Proxy server which allows using multiple SIP accounts with a single SIP login
NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
Net-SIP A Perl SIP framework that includes a stateless proxy
JAIN-SIP Proxy
OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM
OpenSER: GPL SIP Server with TLS support - renamed to Kamailio
OpenSIPS forked from OpenSER.
partysip
SaRP SIP and RTP Proxy in Perl
sipd SIP Proxy
SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org
Siproxd SIP and RTP Proxy
SIPVicious tool suite: tools for auditing sip devices
sipX The SIP PBX for Linux: Complete, native SIP PBX solution from SIPfoundry
Vocal SIP softswitch with H.323 and MGCP translators for non-SIP endpoints
Yxa: Written in the Erlang programming language
SIP Clients (UA's)
Linux clients:
Cockatoo
Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
JPhone Rich software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
Kphone
Linphone audio and video SIP softphone for Linux and Windows XP
minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
OpenZoep: GPL telephone and IM messaging client engine
Peers Minimalist SIP softphone written in java (tested on linux and windows)
PhoneGaim
PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
SFLphone, open-source multiplatform multi-protocol VoIP client
Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
sipXphone from SIPfoundry, previously known as the Pingtel phone
Twinkle
YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
MacOS X clients:
FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
SFLphone, open-source multiplatform multi-protocol VoIP client
Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
Windows clients
Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
JPhone Rich software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
Linphone audio and video SIP softphone for Linux and Windows XP
minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
OpenZoep: GPL telephone and IM messaging client engine
Peers Minimalist SIP softphone written in java (tested on linux and windows)
PhoneGaim
PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
SIP COMMUNICATOR Java based softphone
SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
sipXphone from SIPfoundry, previously known as the Pingtel phone
VMukti (formerly 1videoConference) alpha: a web2.0 VoIP video conferencing software for Asterisk.
wxCommunicator Windows softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support
YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
SIP tools
Callflow: Generates SIP Call Flow diagrams
Open Source Asterisk AMI: Open Source Asterisk AMI interface application
pjsip-perf: SIP transaction and call performance measurement tool
miTester for SIP: SIP testing tool; Automates test execution.
PROTOS Test-Suite: SIP Testing tools
SFTF: SIP Forum Test Framework - a SIP UA test suite primarily targeted at UA software developers hosted by SIPfoundry
SIP-CallerID: SIP Caller ID retrieval and lookup
SIPbomber: SIP proxy testing tool
Sipp: SIP performance tester
Sipper: SIPr (called Sipper) is an open source and a comprehensive SIP application testing framework. Generate any call flow in minutes.
SIP Proxy: SIP security testing tool.
Sipsak: SIP testing tool
SIP Soft client: Software development kit for SIP Softphone
SIPVicious tool suite: tools for auditing SIP devices
SMAP: Locating and fingerprinting remote SIP devices
Vovida.org load balancer: SIP Load Balancer
SIP Protocol Stacks and Libraries
Aloha Spring based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models
eXosip - eXtended osip library
libdissipate SIP stack
minisip includes a SIP stack
MjSip - complete and powerful java-based SIP library for both J2SE and J2ME platforms.
NIST SIP Various SIP appications and tools in Java
oSIP Library SIP Library
OSP client protocol stack and SIPfoundry
PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity
PJSIP: Small footprint, high performance, and ultra-portable SIP stack written in C, and has language binding for Python. Works on Symbian and support ZRTP encryption.
reSIProcate SIP stack and sample Application from SIPfoundry
Twisted Python protocol stacks and applications includes SIP support
Open Sip Stack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry
http://sofia-sip.sourceforge.net Sofia-Sip is SIP stack implementation with STUN and presense support
Verona - GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X
Vovida SIP Vovida SIP stack
YASS - Statefull SIP stack used in Yate written in C++ usable for client, server or proxy in a multithread or single thread model. It's working on both Windows and Linux, it's very small but full featured.
Zocol Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
H.323 Clients
Linux clients:
Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
FreeSWITCH: Console client using OPAL
GnomeMeeting
YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
MacOS X clients:
FreeSWITCH: Console client using OPAL
ohphoneX
Windows clients:
Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
FreeSWITCH: Console client using OPAL
OpenPhone
YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
H.323 Gatekeeper
GNU Gatekeeper - for Linux, Windows, Mac etc.
IAX clients
IAXComm for Linux, MacOS X and Windows
FreeSWITCH
Kiax - for Linux, Windows and MacOS, based on iaxclient, GPL
MozIAX
QtIax from http://www.holgerschurig.de/qtiax.html
SFLphone, open-source multiplatform multi-protocol VoIP client (IAX support is planned)
YakaPhoneSimple, Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware
YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
RTP Proxies
AG Projects: MediaProxy 1 works with SIP express router and OpenSER, has load-balancing using DNS SRV records and accounting capabilities
Maxim Sobolev's RTPproxy: Works with SIP express router to traverse NAT, also functions as RTP gateway between IPv4 and IPv6
MediaProxy 2 is more scalable using kernel space switching and works with OpenSIPs
RTP Protocol Stacks
Secure RTP - see: SRTP
ccRTP C++ library based on GNU Common C++
JRTPLIB C++ object oriented RTP library
libRTP part of gnome-o-phone
libzrtpcpp - ZRTP extension library for ccRTP stack
LIVE.COM Streaming Media includes C++ RTP stack
oRTP Written in C, running on linux, win32 and arm-linux.
PJMEDIA: Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment
RTPlib C library
sipXmediaLib RTP + audio bridges, audio splitters, echo suppression, tone from generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from SIPfoundry
UCL Common Multimedia Library includes cross platform RTP stack
Vovida RTP Stack
YRTP - Yate RTP stack, that can be used in other projects.
Zocol Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
zrtp4j - ZRTP stack for Java, based on GNU ZRTP, used in SIP Communicator
MSRP Relays
MSRPRelay from AG Projects
XCAP servers
OpenXCAP from AG Projects
Other tools
Howler Technologies - optimised G.729 codec for softswitch market.
MORCC - automated online Calling Card store. Paypal integrated.
Oreka capture and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with audio compression, rdbms metadata storage and web based user interface.
Vovida.org STUN server: A STUN server
Voipong - Voice over IP (VoIP) sniffer and call detector.
Vomit converts a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file
PBX platforms
Some of these include SIP proxy functionality
Asterisk: Open Source PBX. Supports IAX, SIP, MGCP, H.323 and other protocols
CallWeaver: a fork of Asterisk with T.38 termination
FreeSWITCH Open Source PBX and Soft Switch
OpenPBX: Open Source PBX developed using Perl
PBX4Linux: ISDN PBX with H.323 GW
SIPexchange PBX Pingtel's SIP PBX
sipwitch: GNU project's Pure SIP call server, sipwitch on freshmeat.net
sipX - The SIP PBX for Linux from SIPfoundry, sipX on freshmeat.net
YATE Yet Another Telephony Engine - supports H.323, SIP, IAX, PSTN
IVR platforms
Asterisk: Open Source PBX with built-in IVR server
Bayonne: GNU project IVR server
CT Server Perl based Open Source client/server library supporting Voicetronix Telephony hardware.
FreeSWITCH
OpenVXI: Implementation of VoiceXML
SEMS: Free/Open Source SIP media server with IVR capabilities
sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
YATE Yet Another Telephony Engine
See Also: VoiceXML
Voicemail servers
Asterisk: Open Source PBX with built-in Voicemail Server
FreeSWITCH
Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
OpenPBX: Open Source PBX with built in voicemail
OpenUMS: Linux Voicemail and Unified Messaging Server
SEMS: Free/Open Source SIP media server with built-in Voicemail and Voicebox Server
sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
VOCP: A Voicemail Server for voice modems
YATE Yet Another Telephony Engine with H.323, SIP and IAX support.
Speech
Text-to-speech and speech-to-text (voice recognition)
Festival: Voice synthesis system (implemented with a trainable neural network)
OpenSALT: Implementation of SALT
OpenVXI: Implementation of VoiceXML
Sphinx: speaker-independent speech recognizer
UniMRCP: cross-platform MRCP client and server
Fax Servers
Asterisk Fax Email Gateway
Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
Hylafax
Development platforms, protocol stacks
H323plus: Open Source H.323 Protocol Stack following on from the original openH323
OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems,
OpenMGCP: Open Source MGCP Protocol Stack Developed with C and POSIX APIs,
OpenSS7: SS7 Protocol Stack
ooh323c: Open Source H.323 Protocol Stack Developed in C
++Skype C++ library for skype add-on platform independent software development. It is platform independent, easy to use, and easy to extend because of the flexible library design, inspired by modern C++ design ideas. Performance is one of the goals.
Radius Servers
Aradial: Radius server and Billing for VoIP
BSDRadius: Radius server for VoIP
Interlink RADIUS Server RADIUS Server Software
RadBox RADIUS Server + Billing System. (For a work, you nead instal Framework 2.0)
Ref : http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software
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