A quick introduction to Kamailio - the leading Open Source SIP server (based on OpenSER and SER). Kamailio is quite different than Asterisk, FreeSwitch and many other VoIP platforms - why is that and how do you start getting your head around Kamailio?
Building VoIP service now, for tomorrow - By Doug HillVoiceSA
In today's business world it is vitally important to develop service based on products that will carry us for the next 10 years. Learn how to deploy IP based voice services now and for the future.
Telephone Wreckers tells you all about Asterisk phone systems - the benefits, features and what product you'll need to build your own custom IP phone system.
Adaptation of a presentation I gave at AstriCon 2007, describing how to use Asterisk as a SIP media gateway in a softswitch, including how to address the problem of optionally supplying the outbound leg with early media.
A quick introduction to Kamailio - the leading Open Source SIP server (based on OpenSER and SER). Kamailio is quite different than Asterisk, FreeSwitch and many other VoIP platforms - why is that and how do you start getting your head around Kamailio?
Building VoIP service now, for tomorrow - By Doug HillVoiceSA
In today's business world it is vitally important to develop service based on products that will carry us for the next 10 years. Learn how to deploy IP based voice services now and for the future.
Telephone Wreckers tells you all about Asterisk phone systems - the benefits, features and what product you'll need to build your own custom IP phone system.
Adaptation of a presentation I gave at AstriCon 2007, describing how to use Asterisk as a SIP media gateway in a softswitch, including how to address the problem of optionally supplying the outbound leg with early media.
Introduction to VoIP, 2nd chapter of "Unified Communications with Elastix" Vol.1
We recommend to read the chapter along with the presentation.
http://elx.ec/chapter2
As you know we promoting the Open Source so in sequence now we have implemented the VOIP in college with the help of Open Source Softwares. You can also do this on your space. We personally ready to assist you.
A presentation about new functionality in SIP that is really needed for Hosted PBX services, SIP on mobile phones and more situations. #SIP #Kamailio #Asterisk #TLS #MoreCrypto
A video with this presentation is available on YouTube at
https://www.youtube.com/watch?v=uqFNlqB_Ssw
The migration from analog or ISDN to Voice-over-IP can lead to significant changes in business communications. For private customers, the migration is generally easy to implement. Many companies, however, face far-reaching changes. Join us on our journey into the future of IP communications.
This presentation contain basic knowledge about how voIP work and what are the security threat in voIP. It will also contain how we can prevent attack on voIP system.
Since the past decade VoIP is making telecommunications cheaper and affordable. In this presentation I have shared some basic concepts of VoIP and Asterisk, the world's best Open Source Software PBX
Introduction to VoIP, 2nd chapter of "Unified Communications with Elastix" Vol.1
We recommend to read the chapter along with the presentation.
http://elx.ec/chapter2
As you know we promoting the Open Source so in sequence now we have implemented the VOIP in college with the help of Open Source Softwares. You can also do this on your space. We personally ready to assist you.
A presentation about new functionality in SIP that is really needed for Hosted PBX services, SIP on mobile phones and more situations. #SIP #Kamailio #Asterisk #TLS #MoreCrypto
A video with this presentation is available on YouTube at
https://www.youtube.com/watch?v=uqFNlqB_Ssw
The migration from analog or ISDN to Voice-over-IP can lead to significant changes in business communications. For private customers, the migration is generally easy to implement. Many companies, however, face far-reaching changes. Join us on our journey into the future of IP communications.
This presentation contain basic knowledge about how voIP work and what are the security threat in voIP. It will also contain how we can prevent attack on voIP system.
Since the past decade VoIP is making telecommunications cheaper and affordable. In this presentation I have shared some basic concepts of VoIP and Asterisk, the world's best Open Source Software PBX
The SlideShare 101 is a quick start guide if you want to walk through the main features that the platform offers. This will keep getting updated as new features are launched.
The SlideShare 101 replaces the earlier "SlideShare Quick Tour".
As humans, we never fail to think that we are highly intelligent beings, and that we are mentally superior than any other creatures found on Earth.
Well, that...... may be true.
However, we can be equally stupid and dumb too.
Worse still, we don't even realize it - in terms of how we can make erroneous judgments, decisions and choices, based on how our mind processes and filters information, as well as how our belief system works.
As intriguing and exciting this topic is to me, I find it difficult to illustrate the concepts involve, and that took me nearly 6 months to complete this work. (The Planning Fallacy in play?!) Throughout writing this deck, I've made a total of 8 major revisions before coming to this final piece.
I hope you'll find this deck both interesting and useful!
Asterisk is an Open Source PBX - but how does it support larger installations? Can you scale it up to thousands of users, with hundreds of simultaneous calls? What about failover, backups and the famous blinking lamps? Olle Johansson goes through various models and describes where some of his current projects with strange names - Pinefrog, Pinana, Pinetree and Bufo fits into this picture.
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
This presentation was intended to present theoretical and experimental concept during setting up a VOIP Server with the well known open source VOIP Server Asterisk@home. Here we have implemented GnuGk PBX and at the last phase we have configured Asterisk server. This was presented at ICACT2007.
SIP servers on embedded systems: Powering SoHo communicationsRADVISION Ltd.
Increasing number of small offices/home offices (SoHo) migrate to VoIP communication based on commodity appliances that provide the connectivity solution.
The appliance is in many cases a combination of:
* Residential gateway (RGW)
* Integrated access device (IAD)
* Key-system/ small PBX
* SIP firewall / GW
In some cases it even can be as part of a set-top box.
All the above is based on SIP server technology that is tailored to embedded devices. This technology involves significant business potential as well as technology challenges that will be elaborated on the webinar.
Positive Hack Days. Gritsai. VOIP insecurities workshopPositive Hack Days
Участник получит представление об основе IP-телефонии, а также базовые навыки поиска уязвимостей на примере распространенных IP-PBX и абонентских устройств. Рассматриваются как типовые сетевые уязвимости, так и сложные случаи, обнаруживаемые в ходе анализа защищенности реальных сетей.
Kamailio is the leading Open Source SIP Server - a SIP proxy, registrar, location server, presence server, IMS server and much more. Find out more by viewing this quick presentation! (Updated June 2014)
10. 200 OK SIP From: ‘Graham Francis’ <sip:6886756@sipcom.com> To: <sip:5767445@sipcom.com> SDP Let’s use G711…!
11. ACK SIP From: ‘Graham Francis’ <sip:6886756@sipcom.com> To: <sip:5767445@sipcom.com> SDP OK, G711 it is….!
12. ACK VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP VoIP SIP From: ‘Graham Francis’ <sip:6886756@sipcom.com> To: <sip:5767445@sipcom.com> SDP OK, G711 it is….! VoIP
13. SIP SERVERS SIP Registrar SIP Proxy Location Service REGISTER 200 OK Database Hi, today you can find Extension 1001 @ 192.168.100.77 INVITE 1001 where are you?
25. 1 x Broadband Connection 2 x SIP trunks Service Provider PSTN PBX
26. PBX Router PSTN PSTN ITSP Offerings We need some more DDI’s Can I have a ‘London’ number? No problem Can we have our ‘old’ numbers please? Of course! That’s us
27. PSTN ITSP Offerings PBX Router SHA / TLS / SRTP Thwarted PSTN We need more lines more our marketing push! I can do that via our Web Interface I’ll do that!
49. AT&T BBC Cable and Wireless Carousel Industries Cedarpoint CenturyLink Cisco Deloitte Ingate IPC Level3 Microsoft Mitel Mtsallstream Nacr Nec America Polycom Shared Technologies Telematrix Telquest Verizon Virtualhold West Windstream Xeta
50. The Telecommunications Industry Association (TIA), the leader in advocacy, standards development, business development and intelligence for the information and communications technology industry, has officially endorsed the The SIP School as the provider of choice for training and certification for Session Initiation Protocol (SIP). AT&T BBC Cable and Wireless Carousel Industries Cedarpoint CenturyLink Cisco Deloitte Ingate IPC Level3 Microsoft Mitel Mtsallstream Nacr Nec America Polycom Shared Technologies Telematrix Telquest Verizon Virtualhold West Windstream Xeta
Start immediately, here’s the USB, fire up and install X-Lite + Wireshark
and it’s job is to establish connections between end devices for things like Voice and Video communications. Take a look at these two
So, with SIP It’s job is to establish a connection through signalling. Here you can see an IP sending a SIP Invite to call another device along with some codec to select
The device responds and even picks a codec
After a final acknowledgement is set up, media is transmitted And I know that there are protocols out these that already do this, but the special thing about SIP is that it’s both truly open as no-one owns it and being based on the HTML standard it opens up a whole world of interoperability with other applications on your network which you are seeing demonstrated at this conference.
After a final acknowledgement is set up, media is transmitted And I know that there are protocols out these that already do this, but the special thing about SIP is that it’s both truly open as no-one owns it and being based on the HTML standard it opens up a whole world of interoperability with other applications on your network which you are seeing demonstrated at this conference.
A location server will have a database to hold location information for SIP User Agents and this is kept up to date via the Registrar Service that is processing Register and Re-Register messages from clients. The SIP proxy will then use the Location service to help find the current location details of the destination SIP URL in a SIP INVITE message. Once it has it’s answer it can forward the INVITE to the destination and tell the Calling device that it is trying. Again, there are many developers and providers out there deploying SIP and you may find that all of these service can run on one single server.
Student 1, makes call to Instructor phone
So, let's make it crystal clear. A “SIP trunk” uses either new or existing broadband Internet connectivity to supplement or replace traditional telephone company circuits connecting an organization's telephone systems to one another and to the outside world. SIP trunks are not a “trunk” in the old telephone company sense in that they are not a “hard-wired” circuit. Rather, they use the miracle of the Internet to originate or terminate telephone calls to and from the Public Switched Telephone Network (PSTN) or other Internet-connected locations. In reality, a SIP trunk is actually a concurrent call using the Internet pipe for access and egress.
Well, your selected ITSP should be able to provide you with Number Ranges, Move your existing numbers to their network, provide Emergency services support and Also have a good technical support team just in case you need help Features all there?
Also, they need a good network infrastructure to ensure continuity of service… Give you a choice of Codecs to use and be able to provide a Secure connection to their network. And when you need more Trunks, can you manage your settings or do they do it for you? Security ITSP implement SIP security? RTP Security? SBC Decrypt after SBC? New Stack on PBX / IP Phones Firewalls etc.
If the PBX does not have the capabilities to apply security mechanisms it’s wise to use either a ‘SIP aware’ edge device to set up the TLS session with the ITSP or direct all SIP traffic to a Session Border controller at the ITSP which will then secure your signalling. It’s also important to remember that between your edge and the ITSP only mechanisms such as SRTP and IP-Sec will secure the actual Audio or other media transmitted.
‘ I’ call an outside line (remember I need to trace) Voip info number or a Cell of Student? Show Trace / breakdown Can I play it back? Do I need Headphones or USB Headset?
SIP Workbench
Filtering Colouring Rules Export to other tools?
Now we have a lot of friends using our system already, including some pretty good endorsements of our training and SSCA Certification. And today I can tell you about our new endorsements which in effect make us the leading SIP training and certification company – worldwide.
Now we have a lot of friends using our system already, including some pretty good endorsements of our training and SSCA Certification. And today I can tell you about our new endorsements which in effect make us the leading SIP training and certification company – worldwide.
Welcome, my name is Graham Francis – I’m from The SIP School and thanks for coming along today “ Understanding SIP because you’ll need to” is all about looking at what we’ve been used to such as analog / digital telephone systems and what we’re migrating to ~ VoIP and SIP networks … Why do you need to understand SIP? Well. It’s the foundation for all solutions but it may never be finished as a protocol so the ground is always moving under your feet so you’ve got to be ready But let’s start with the basics