Comments on: VoIP Over VPN: Securely Interconnecting Asterisk Servers

By: Ayesha

Ayesha — Fri, 29 Nov 2013 04:33:44 +0000

Hay, what should we do if we centralize one asterisk pbx?
Forexample:
A, B and C are three asterisk Pbx.
A is centerlize asterisk box and A is connected to both B and C through sip trunk. If B wants to talk to C, it will be route through A..

By: Nerd Uno

Nerd Uno — Mon, 10 Dec 2012 01:41:33 +0000

For another approach to safely interconnecting Asterisk servers without the hassle of managing dynamic IP addresses or VPNs, see this thread for tips on implementing UserAgent Knocks with iptables.

By: Imran Malik

Imran Malik — Tue, 06 Apr 2010 10:59:44 +0000

Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.

Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.

Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.

By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.

How VoIP Works

When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.

What are speech codec’s and what role codec plays in VoIP?

Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.

Following is a list of VoIP codec’s along with how much data network bandwidth they consume.

* AMR Codec
* BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
* GIPS Family – 13.3 Kbps and up
* GSM – 13 Kbps (full rate), 20ms frame size
* iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
* ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
* ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
* ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
* ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
* ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
* ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
* ITU G.726 – 16/24/32/40 Kbps
* ITU G.728 – 16 Kbps
* ITU G.729 – 8 Kbps, 10ms frame size
* Speex – 2.15 to 44.2 Kbps
* LPC10 – 2.5 Kbps
* DoD CELP – 4.8 Kbps

Switch to VoIP Today and you will never want to use traditional PSTN ever again.

Thanks

-Imran

By: Mike

Mike — Mon, 22 Jun 2009 02:20:52 +0000

Ward,

I can only drool over the thought of having a home in Balsam Mountain. Being a descendant of a Cherokee Indian makes me want to move to the mountains as well.

By: Mahmood

Mahmood — Sun, 14 Jun 2009 15:58:36 +0000

Currently we use 512Kbit/s B/W VPN service from VPN4VOIP company together with our Asterisk in PK to talk with our Asterisk PBX in US. Works pretty well. They reroute US IP to our PK PBX so make things are much simpler for us. Someone may wish to try this implementation, too.

By: Eduardo Silva

Eduardo Silva — Fri, 12 Jun 2009 07:13:21 +0000

Thanks for the "free tip of the week"!! Proxmox is great. If you do have an openvz template of piaf and want to share, I would like to use it 🙂

By: Jan

Jan — Thu, 11 Jun 2009 21:06:38 +0000

During CeBIT’09 in Hannover, Germany we introdused a new version of our firmware for the IPxx series of PBX’s. Our new firmware scheduled to be released within the next few weeks is called VoIPtel SEq and has an integrated OpenVPN server. During CeBIT we demonstrated secure phonecalls between two SNOM 370’s passing through an IP01, a PBX the size of an ATA capable of 34 concurrent calls. The VoIPtel SEq firmware will make it easy to set up both secure lines as well as secure trunks between PBX’s.

The IPxx are small and inexpensive embedded devices using the Blackfin DSP as the only processor. Both hardware as well as firmware is open source.

By: Skavoovie

Skavoovie — Mon, 08 Jun 2009 18:55:29 +0000

Signed up for Sipgate free account a few minutes prior to this post. Got the confirmation email just fine and got verified, but they only have phone numbers in California 415 area code right now, so skipped adding a number to my account. If they do what they claim, I’ll add a local area code number and give them a try. Til then, not even bothering to add the trunk.

By: Jerry

Jerry — Sun, 07 Jun 2009 20:40:45 +0000

Has anyone had problems signing up for the free Sipgate account? I registered and clicked "send e-mail" a couple of times but never got the registration e-mail.

By: Tim Henning

Tim Henning — Wed, 03 Jun 2009 20:31:45 +0000

In the article VoIP Over VPN: Securely Interconnecting Asterisk Servers, you are using IAX trunks between the PBX servers. Why are you not using DUNDi? This provides a cleaner more manageable way to interconnect boxes and share resources. Just Curious.

[WM: Coming soon. Stay tuned.]

By: Sidimustafa

Sidimustafa — Wed, 03 Jun 2009 20:06:23 +0000

Hello Ward.
I agree with Andy, but not perse about the licensing issue, but more about the 3rd party involved.
Should Hamachi’s network has a hickup, that’s a next complex in the already complex network (internet). That means, you’ll be down until Hamichi is back up and running (remember skype – we never expected them to be out for 3+ days)…
I have servers connected via OpenVPN, of which i took the writeups from the pbxinaflash forums..

On many occasions i saw your commented about openvpn, but still champion Hamachi in many of your articles.

Is there a special reason you champion Hamachi? OpenVPN vpn is just as easy to setup, can all be done via script also.
OpenVPN also has little more features than Hamachi, mainly that u can connect to the entire Network, hamachi can only connect to the PC it’s install on.

Think about this, you have a client, servers connected via OpenVPN, you connect to them via that server, VoIP telephones are on the same subnet as the OpenVPN, this means, via that VPN connection, you can route traffic to the VoiP phones, and http to them, and no need to have access to the Total Network (very good, if you only have the PBX contract, and not the contract for the entire network)

By: Andy Lauppe

Andy Lauppe — Tue, 02 Jun 2009 19:17:15 +0000

I know you've fielded this concept endlessly since PiaF started playing with VPN's, but with Hamachi's purchase by logmein and "free only for personal use" policy with limited endpoints, would you be willing to consider OpenVPN if someone wrote a decent how-to specifically for PiaF machines? Hamachi is now so 'closed', it gives even the most liberal 'open source' people the willies... [WM: We'd love to see a good writeup of OpenVPN. We were unaware of the license changes in the newer Hamachi versions to which you refer. The version of Hamachi (0.9.9.9) that we recommend is available from thousands of different download sites and contains no such restriction. But thanks for bringing 'the change' to our attention. And thanks for the OpenVPN offer. We accept.] The Hamachi 0.9.9.9 version available from many Linux download sites as well as LogMeIn's own web site (which redirects to the original developer's site) includes the following license:
1. License. Applied Networking grants the End-User a limited, non-exclusive, nontransferable, royalty-free license to install and use the Client including software, documentation and any fonts accompanying this License whether on disk, in read only memory, on any other media or in any other form.
I just downloaded the 0.9.9.9 version (from the LogMeIn link above which is freely available on Google) moments ago so it appears LogMeIn has decided not to apply the same licensing restrictions to version 0.9.9.9, even assuming they could.