Build your own VoIP System Part 1: The Basicsdocshare01.docshare.tips/files/12552/125527000.pdf · 2016-06-02 · Build your own VoIP System – Part 1: The Basics by Andreas Granig

Build your own VoIP System – Part 1: The Basics

by Andreas Granig on March 17, 2012 · 5 Comments">5 Comments · in Technical

This blog post is the first part of a series of posts, which describe how VoIP works and how the Sipwise

sip:provider Platform enables you to start various VoIP business models.

Part 1, which is provided in this post, gives you an introduction in how VoIP works.

Part 2 shows how you can set up a secure and self-hosted Skype-like VoIP system for free using the

sip:provider Platform within 30 minutes.

Part 3 is dedicated to the sip:provider Platform acting as an SBC in front of existing VoIP systems.

Part 4 describes how you can operate a whole-sale business with the sip:provider Platform.

Part 5 shows how to enable Over-The-Top (OTT) services using Apple and Google Push

Notification Services.

Introduction

VoIP Systems are seen as complex communication infrastructures even from a high level perspective, but

they‟re not. Well, VoIP is in fact complex in its details, but it has been abstracted by various projects in

order to make it really straight-forward to use it, so it‟s easy to start a compelling voice/video

communication system or service (which I‟ll name “VoIP system” or “VoIP service” throughout the

document) from scratch, but it‟s important to learn a few facts about it in order to choose the right base

system for successfully running a VoIP service.

The Basics

VoIP just means “Voice over IP”, which is a generic term for transporting real-time voice sessions over the

Internet. However, it doesn‟t define HOW this is done, and even the term “Voice” is a bit misleading,

because with the very same concept, you can transport also Video and Fax over an IP connection.

There are a couple of elements involved when you‟re talking about a VoIP system:

To sum it up, there are SIP Endpoints, which are the client instances of your customers. These could be

software installed on your customer‟s computers (popular software is Jitsi, an open source and cross-

platform communications client, or Bria, a commercial multi-platform client for Windows, iOS and

Android). Other possibilities are SIP phones like SNOM phones or Polycom Phones.

Beside the customer facing end points, there are SIP gateways which translate VoIP into traditional fixed-net

and mobile networks. They pretty much act like customer facing clients, but usually are able to handle

multiples of parallel calls. They are usually connected via multiple ISDN E1 or T1 lines, and sometimes an

SS7 control layer is used on top.

How does SIP work?

In order to establish a communication session, you need a signaling protocol, which tells the involved parties

who want to communicate with whom, and which media capabilities might be used (e.g. plain voice,

voice/video, fax etc.). There are several protocols out there, like Skype (a proprietary protocol) and H.323

(more or less obsolete since 2004) and the most important and nowadays most wide-spread one and the one

we‟re concentrating here: SIP, the Session Initiation Protocol.

SIP Registrations

A very important part of VoIP is the registration of customer endpoints. It means if a customer starts its SIP

client, the client tells the SIP server at which IP and port it is reachable in case there‟s a call towards this

customer. The call flow looks like this:

The important part, beside the authentication scenario which is a http digest authentication, is the Contact

header, which indicates at which IP: port the customer is reachable.

So during start-up, the client tells the server the contact address it‟s reachable for subsequent calls.

But what about real Phone Numbers?

Ok, so we learned that [email protected] can contact [email protected] if [email protected] registered

up-front (telling the SIP service provider at which IP: port he‟s reachable), and vice versa. But, what about

real phone numbers?

In order to receive calls from the PSTN (public switched telephony network), your SIP service provider

needs to map a PSTN number to your SIP URI, e.g. he needs to know that [email protected] is an

equivalent to for example +43 1 1001. If somebody calls 4311001 in the PSTN, it‟s routed through the

telephony network down to your service provider, which holds ownership of that number. The service

provider is now responsible to translate the number to a corresponding SIP-URI, and then route the call to

the IP:port where this user is registered.

How about a Video Stream?

The important thing here is that any media stream in SIP signaling is negotiated end-to-end. This means that

if [email protected] calls [email protected], alice proposes a list of media sessions (e.g. audio with a

specific list of codecs, and video with a specific list of other codecs), and bob compares this list with its own

capabilities and then replies with a (potential) sub-set of the offer from alice. So if alice proposes an audio

and video call, but bob doesn‟t have a web-cam, he‟ll reply with a sub-set of alice‟s offer, which only

contains the audio-part. However, if bob has a web-cam, we‟ll reply with an according answer telling alice

that both audio and video streams are available.

Conclusion

If you want to work with “VoIP”, you most likely will work with the SIP Protocol. SIP will allow you to do

two-way, end-to-end communication, but you‟ll need SIP clients to attach to a system like this. Do you need

do pay for an external service in order to start a VoIP system? No!

What’s next?

The next post will describe how you can use the open source Sipwise sip:provider CE to build a VoIP

system from scratch within an hour. It‟ll show how you can create a Skype-like service within your network

using IPv4, IPv6, TLS and SRTP.

Build your own VoIP System – Part 2: An open Skype Replacement

by Andreas Granig on September 5, 2012 · 7 Comments">7 Comments · in Technical

In Part 1 of our series “Build your own VoIP System” we learned about the very basics of how VoIP and

SIP in particular works.

This is Part 2, describing the process of setting up a Skype-like service using the sip:providerCE

.

In Part 3 you will learn how to protect existing VoIP deployments with the sip:provider acting as a Session

Border Controller (SBC).

Since version 2.6 of our sip:provider platform, we got everything in our hands to build a secure and self-

hosted Skype-like communication service solely based on open source software.

The Goal

In this post, we attempt to build a free, secure, SIP based communication system to provide encrypted voice

and video communication, buddy lists, instant messaging, presence and remote desktop sharing/control on a

self-hosted system.

Once you‟re done with that, adding skype-in/skype-out features to receive and place calls from/to the

traditional telephony/mobile network is fairly easy, but will be covered in a separate post.

The whole process will take around 30 minutes up to an hour for an initial setup, so grab a coffee and clear

your head.

The Ingredients

For our system to work, we need a communication server and a proper client for our end users.

The Server

As a communication server, we will use sip:providerCE

v2.6. The easiest way to get started with it is to

download the VMware or Virtualbox image and fire it up on a suitable machine. If you get more serious,

you want to install the system from scratch on a dedicated server with a public static IP. If you‟re new to

VoIP and SIP, do NOT try to install it on an Amazon EC2 instance, as they‟re using destination NAT, which

is a big pain for SIP and needs some experience with the SPCE to tweak it properly for that scenario.

Note that the SPCE is a 64bit system, so in order to run the VM images, you need to turn on 64bit CPU

virtualization in your BIOS if VMware or Virtualbox warns you about it.

The Client

Like with Skype, your users will need a Client software to leverage the full potential of the server. The good

thing about a SIP based system is that you can hook up pretty much every SIP client (IP phone, ISDN

adapter, Desktop client, Mobile client) to the SPCE. This usually works fine with just voice/video

communication, but with advanced features like presence, diversity leads to interoperability issues, so the

SPCE server is optimized for Jitsi, a Java based multi-platform client providing all the features we require

for this tutorial.

Install the Server

In our setup, we will use Virtualbox to boot the Virtualbox VM image of the SPCE. Follow these steps to

get started:

1. If you don‟t have Virtualbox 4.x installed yet, download it from here and install it, or upgrade your

older version.

2. Download the Virtualbox VM image of the SPCE from here.

3. Start Virtualbox. On Linux, you can start it like this: $ virtualbox sip_provider_CE_2.6_virtualbox.ova You will be prompted to import the new VM, which will look similar to this:

Once the import is finished, double-check if the network interface is the correct one providing access

to the Internet:

In this case, it shows eth1, where I really want to use wlan0 because this is the interface into my

LAN network I use for testing. To change it, click Settings on the top and change it in the Network

section, like this:

Use the proper interface suitable for you, wlan0 will most likely not work for you!

IMPORTANT: Keep the mode to Bridged Adapter to avoid any NAT on the server side!

4. Start the sip_provider_CE VM instance.

5. Once the login screen is shown, log in with user root and password sipwise.

6. Check your network settings. The VM instance is set to use DHCP by default. If this is fine with you,

execute ifconfig eth0 and remember the IP address of this interface, and continue to the next step.

However if you want to configure a static IP address, you need to edit /etc/network/interfaces, e.g.

like this:

auto lo

iface lo inet loopback

auto eth0

iface eth0 inet static

address 192.168.0.122

netmask 255.255.255.0

gateway 192.168.0.1

dns-nameservers 8.8.8.8 8.8.4.4

Then execute ifdown eth0; ifup eth0 to bring up the interface with the proper configuration. Then edit

/etc/ngcp-config/config.yml, search for eaddress and and change the option to the IP address you

statically set above, like this:

eaddress: 192.168.0.122

7. The last step on the command line is to execute the command ngcpcfg apply to generate the platform

configuration files and reboot the server (only needed for the first time for simplicity reasons to make

sure all services are started correctly).

Configure the Domain and Users

Now that the SPCE system is up and running, point your browser to the Administrative Web Panel located

at HTTPS port 1443 of your IP you configured or got via DHCP above, like this:

https://192.168.0.122:1443

The username is administrator and the password is administrator.

There are a couple of steps to get your first users online:

1. Create a Domain for your users. Your users will have subscribers identified by a so-called SIP URI

like sip:[email protected], and similar to virtual hosts on a web server, you can create as many

domains as you like in order to partition your users. Just make sure that the domain name you define

here is pointing to the IP address of the system. You can also directly use your IP address for testing

purposes, so a user would be [email protected] in my case, and I‟ll use that throughout the rest of

the post.

To create the domain, go to System Administration → Domains, enter your Domain name or IP

address and press the Add button.

2. Create an Account for your user. On the SPCE, accounts are billing containers for one or more

subscribers. Usually one user will have one account with one subscriber in it.

To create an account, go to Accounts → Create new account and press the Add button. You will be

presented with the Billing Settings, which we just keep at its defaults for now, so we press the Save

button.

3. Create a Subscriber within the new account. At the bottom of the Account page is the Subscribers

section. Click the Create button to configure a new Subscriber.

The only mandatory information is the SIP URI and the SIP password fields. If you set the Web User

and Web Password as well, the user is able to log into the Customer Self Care Web Interface running

at HTTPS port 443, e.g. https://192.168.0.122.

4. Repeat steps 2 and 3 for all the users you want to create.

That‟s it on the server side. There is a lot more you can configure and tweak for Domains and for

Subscribers, but it‟s not important for your first tests. There is the SPCE Handbook providing all the detailed

information, which you should check to learn about advanced configuration options.

Install and configure the Client

Your users need to download and install the Jitsi Client. Make sure to use the v1.1 latest Nightly Build. All

the cool features won’t work otherwise (e.g. with the stable 1.0 version line).

The setup process is still quite bumpy compared to setting up a Skype client, because there are some manual

steps involved. We‟re working on that part providing an SPCE-optimized version of Jitsi, but up until now it

works like this:

When starting Jitsi for the first time, you will be presented with a Sign In page. Choose the Use online

provisioning link at the very bottom of this window.

Check the Enable provisioning check box, select Manually specify a provisioning URI and put the following

URL there, with only the IP address part changed to reflect your IP or domain name (make sure to leave

the rest intact exactly as shown here):

https://192.168.0.122/jitsi?user=${username}&pass=${password}&uuid=${uuid}

In case Jitsi was already installed, make sure to click the Forget Password! button as well.

Then exit Jitsi and start it again!

During the next startup, Jitsi will pop up an authentication window asking for your username and password.

Enter your SIP URI and your SIP password here.

.

If you‟re asked for a Certificate Verification, click Show Certificate, select the Always trust the certificate

check-box and click Continue anyway.

To avoid this warning, you have to upload a properly signed SSL certificate to the server and configure it.

Check the chapter in the Handbook to learn how to do.

That‟s it. Jitsi will download the configuration via an SSL encrypted HTTP connection, should register

successfully to the server (might take some seconds) via a TLS encrypted SIP connection, and will fetch the

buddy list (empty for a new user) also via an SSL encrypted HTTP connection.

You can now add other contacts to the buddy list like you know it with other services like Skype. You can

place a voice or video call by calling the username (e.g. bob) if the other party is within the same doman, or

username@domain (e.g. [email protected]) if the other party is within another domain.

Next Steps

Please consult the SPCE Handbook to learn how to configure phone numbers for your subscribers, how to

configure subscriber features like call-forwards, call-blockings etc, and how to add SIP Peerings to connect

to the traditional phone network in order to place and receive calls to/from landlines and mobile phones.

Build your own VoIP System – Part 3: the sip:provider as an SBC

by Jon Bonilla on September 5, 2012 · 3 Comments">3 Comments · in Technical

In Part 1 of our series “Build your own VoIP System” we learned about the very basics of how VoIP and

SIP in particular works.

In Part 2 the setup of a Skype-like service was explained.

This is Part 3, where you will learn how to protect existing VoIP deployments with the sip:provider acting as

a Session Border Controller (SBC).

Introduction

Over the last years, we‟ve been developing the NGCP technology to make our products the best Class5

softswitches in the market. 90% of our deployments are Class5 scenarios, where we do green field

deployment or migrations from legacy systems. There are three ways we‟ve deployed the NGCP based

products though:

As Class5 platforms. This is the usual way as we explain in the Handbook.

As SBC systems, to protect and enhance already existing Class5 systems.

As wholesale systems, just sending calls between peers and doing accounting of high traffic.

Benefits

This blog post explains the SBC scenario and its benefits for already existing, in production Class5 or PBX

systems. Before we start the technical part, let‟s imagine a simple scenario: You have a company with

people around the world and your PBX system is exposed to the internet to grant access to your employees

or to let customers call your company via SIP. You now lock down your PBX system and only allow traffic

coming from a Sip:Provider system that will act as SBC in front of your PBX system and will be the only

element exposed to the Internet. Why would you do that? What does the Sip:Provider provide to the PBX

system?

1. Security:

The sip:provider platform provides DOS protection system, being able to handle thousands of

requests per second, banning the originating IP addresses without measurable performance impact.

It also provides DDOS attack protection to prevent distributed attacks bypass the DOS protection to

bruteforce a subscriber‟s password.

It can limit the number of overall simultaneous calls and outgoing simultaneous calls per subscriber

and per account. In our example, the PBX can do the same but it could be that we want to have

different limits for direct connections to the PBX in the company LAN or for road-warrior

connections.

It can block incoming and outgoing destinations. Maybe we want to restrict the destinations for

outside connections while we have another set of permissions for in-office connections.

It has a real-time billing/rating system with prepaid profiles to limit the amount of

calls/minutes/expense per month (PRO systems only)

It has an anti-fraud system with automated notifications and/or subscriber blocking mechanisms in

case a threshold has been reached in a billing period.

2. Functionality:

It has full UDP/TCP/TLS signaling support. The subscribers can talk TLS with the sip:provider SBC

while it will talk UDP with the PBX/Class5 system. This is something recommended on mobile

devices not only because the security you achieve by encrypting the signaling but also because you

avoid most SIP ALG mechanisms and traffic filtering.

It has full IPv6 support. It can translate IPv6/IPv4 for both media and signaling with its dual stack.

It has parallel forking support. It will handle multiple registrations for the same subscriber, but will

register only once against the PBX.

It has cancel reason support for parallel forking.

It has a reliable real time billing and rating system.

It provides per-leg session timers. You can set different session timers to each leg of the call, or even

disabling it for endpoints that do not support it.

It has RTP timeout detection with automated call teardown.

It provides codec filtering. You can whitelist/blacklist the codecs that will be negotiated between the

endpoints. You can also manipulate codec preferences.

You can set a max call duration for any call and tear it down once the limit has been reached.

It can take care of handling the NAT for the subscribers and sending the keep-alives to each device.

It can use different credentials for external subscribers and for registering against the PBX/Class5.

That way subscribers don‟t need to know the SIP credentials of the PBX system.

You can perform SIP method and header filtering for low level manipulation.

There are early media announcements for several failed call scenarios like a busy or offline callee

etc.

It is multi-domain capable. This means that a single provider can act as an SBC for multiple

Class5/PBX systems at the same time. With that feature, it can perform as a routing hub for multiple

systems.

It is open source, so it can be extended and tweaked even to more exotic needs and use-cases.

3. Performance:

The sip:provider platform is designed to handle thousands of concurrent calls and subscribers. The

sip:carrier platform can handle millions of subscribers. It is a scalable technology that runs on carefully

specified Dell servers for the PRO and on IBM Bladecenters for the Carrier. It is much more cost effective

to scale a sip:provider system than any other systems. In terms of performance let‟s imagine this scenario:

One subscriber has 5 different devices registered against on a sip:provider system. They use TLS for

signaling, they are behind NAT, the registration interval is 5 minutes, the NAT keepalive is 30 seconds,

session timer period is 90 seconds and the client devices also send OPTIONS messages to the server. In the

other leg, the sip:provider registers once against the PBX, with a registration expires of 6 hours, no NAT,

using UDP as transport and session timers of 60 minutes. It‟s clear that the traffic and the system load is

lower in the PBX side without having to directly process the subscriber traffic. If you extend this to

hundreds or thousands (or millions!) of subscribers you‟ll see that the existing system can handle more

subscribers with the same hardware and less PBX/Class5 licenses.

Let’s go Technical

This won‟t be a step-by-step tutorial of deploying a sip:providerCE

system (SPCE) as SBC. We offer

deployment and configuration services as well as training and support contracts for this. Let‟s assume that

you have already read the Handbook of the 2.6 system (the stable version at this point). Although it

describes a Class5 scenario deployment, almost everything needed in order to deploy the SBC scenario is

covered there.

The necessary steps are:

1. Read the Handbook. Really, do it.

2. Install the system

3. Set the networking eaddress in config.yml

4. Create a domain (we‟ll use the IP address as domain in this example)

5. Create an account

6. Create a couple of subscribers

7. Create a peering group with the PBX as gateway and an empty (match everything) peering rule for it.

Let‟s use examples:

Please note that for this deployment we won’t touch the already-in-production system configuration.

Asterisk system: We‟ll use E164 numbers as username to simplify the example. We can always use rewrite-

rules in the Provider or in the PBX to work with different numbering plans. I‟ve set up a 1.4 version since

it‟s a very common version in production environments. It would work the same way with other versions or

even distributions such as Elastix.

IP: XX.XX.XX.XX

Subscriber 1: 4312001

Subscriber 2: 4312002

Sip:Provider SBC system:

IP: YY.YY:YY.YY

Domain: YY.YY.YY.YY

Subscriber1: Testuser1

Subscriber2: Testuser2

Peer group: Asterisk PBX; Peer gw: XX.XX.XX.XX; Peer rule: Blank

We‟ll make some changes to the low level configuration. We‟ll disable extra services and rating as we are

using the most simple scenario in this example. Other changes like language-, address- and domain-settings

are omitted.

asterisk.voicemail.enable: „no‟

kamailio.proxy.allow_non_numeric_to_pstn: „yes‟

kamailio.proxy.presence.enable: „no‟

networking.eaddress: YY.YY.YY.YY

rateomat.enable: „no‟

sems.conference.enable: „no‟

sems.vsc.enable: „no‟

After modifying the file, apply the changes using the configuration framework: ngcpcfg apply

Now we can deal with the high level configuration settings. As I mentioned before, we‟ll create the

YY.YY.YY.YY domain in the SPCE Admin panel. We‟ll enable some default preferences in this domain

that will be the default for every subscriber. Remember that any domain preference will be overwritten by

the subscriber preference if it is set there as well. For the purpose of this example just one preference is

required: “force_outbound_calls_to_peer“. This setting will force all calls to SPCE subscriber to be routed

to the peer even if the callee is local.

Create subscribers with proper SIP URIs and passwords. End-users will require those credentials to

provision their softphones, tablets, mobile devices etc. After the master data is filled and the user is created,

we need to tell the system to register on behalf of the subscriber to the PBX system. That‟s easily done by

just filling the preferences “peer_auth_user”, “peer_auth_pass”, “peer_auth_proxy” and

“peer_auth_register” with the PBX credentials.

Once the preferences have been saved, the SPCE will automatically send a REGISTER request to the PBX

system:

And that‟s pretty much everything. The PBX will see the subscribers registered exactly as if there was no

SBC in between so no changes need to be made to the configuration. Meanwhile, the SPCE will handle the

attacks, NAT, transport and protocol translation, keepalives etc.

There‟s still some work to do. Enabling REFER to be sent to the PBX (this is disabled by default in the

SPCE) is needed if you put the SPCE in front of a PBX and phones use REFER for call-transfers. It is not

needed in front of Class4 or Class5 switches. To enable that, it requires white-listing the REFER method in

kamailio-proxy and white-listing some headers in the sbc profile.

Questions, Comments and Feedback

If you have any questions regarding the deployment, you can always use our support mailing list or contact

“sales _AT_ sipwise.com” for professional services.

Remember that sip:providerCE

is 100% free. You can install and use it without limitations. So, go and do a

test deployment!

Enjoy!

Discussion

Community

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Maurice Madrid • 12 days ago

Excellent! really!! but...where is the other part? hehehe

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sipwise Mod Maurice Madrid • 12 days ago

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Part 2 and 3 are linked in the first paragraph. 4 and 5 are not ready yet.

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Craig • 2 months ago

This is great!!!!!

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Craig • 2 months ago

This is great!!!!!

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Craig • 2 months ago

This is great!!!!!

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als7ly • 4 months ago

hi ,

is this mean that i can develop a desk top software that works and present exactly the features of messengers

like yahoo msn beylexe

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Hello • 6 months ago

Thanks a Lot for giving us great prodct

is there anybody one problem can help me to solve.. i cant login the provission

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William Ikiabo • 10 months ago

Thanks for this article, I can't wait for the complete series.

Keep on the nice job

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William Ikiabo • 10 months ago

Thanks for this article, I can't wait for the complete series.

Keep on the nice job

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Ismail • 2 months ago

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Great post, is there a way to make an installation on the ec2 or rackspace cloud?

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sipwise Mod Ismail • 2 months ago

Rackspace should be straight forward, just follow the handbook how to set up a CE on an

ordinary server. Maybe you can even import the vmware image. EC2 is a bit more tricky,

check our mailing list how to use the aaddress setting.

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Ismail • 2 months ago

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Great post, is there a way to make an installation on the ec2 or rackspace cloud?

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Ismail • 2 months ago

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Great post, is there a way to make an installation on the ec2 or rackspace cloud?

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Jon Bonilla • 2 months ago

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Please, for technical questions, refer to our mailing list.

http://lists.sipwise.com/listi...

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alex • 2 months ago

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I love the tutorial and was able to set up my server, but I am wondering how i can port forward this

in my router to access this off my LAN. what port does this go through? 1443?

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alex • 2 months ago

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I love the tutorial and was able to set up my server, but I am wondering how i can port forward this

in my router to access this off my LAN. what port does this go through? 1443?

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skin whitening product • 4 months ago

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hi!,I love your writing very a lot! share we keep up a correspondence more about your article on

AOL? I require a specialist in this house to solve my problem. Maybe that's you! Having a look

ahead to look you.

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Mike Telahun • 5 months ago

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I spent a day and a half banging my head against these two issues:

1. When doing the manual provisioning in the jitsi client in the URI field pass=${pass} should read

pass=${password}. Otherwise, when you restart jitsi it will not bring up the username/password

dialog.

2. After I got past the first issue I could make and receive calls fine, but my contacts in the contact-

list were showing as off-line. For some reason jitsi was not passing my sip credentials on to the

presence server. To fix this:

a. Tools -> Options

b. In the Accounts tab select the SIP account it created for you and click the Edit button at the bottom

c. In the Presence tab un-check the "Use SIP Credentials" check box and manually enter your

username and password.

Other than that, Great Work!

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Mike Telahun • 5 months ago

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I spent a day and a half banging my head against these two issues:

1. When doing the manual provisioning in the jitsi client in the URI field pass=${pass} should read

pass=${password}. Otherwise, when you restart jitsi it will not bring up the username/password

dialog.

2. After I got past the first issue I could make and receive calls fine, but my contacts in the contact-

list were showing as off-line. For some reason jitsi was not passing my sip credentials on to the

presence server. To fix this:

a. Tools -> Options

b. In the Accounts tab select the SIP account it created for you and click the Edit button at the bottom

c. In the Presence tab un-check the "Use SIP Credentials" check box and manually enter your

username and password.

Other than that, Great Work!

Samir Kundu • 14 days ago

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Who do I turn to get help to install and configure , so as

to run VoIP class 5 soft switch?

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Cal Leeming • 3 months ago

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lol asterisk.

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Ali Ardestani • 5 months ago

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With this config once the phone on PBX is not answered the SBC keeps retrying and never goes to

voicemail

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Jon Bonilla Ali Ardestani • 4 months ago

You can configure a ringtimeout in your subscriber. For technical questions, please, go to our

mailing list.

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