One such SIP compliant softphone is called 3CXPhone. This is a simple client that will work over both WiFI and 3G.

Bandwidth!! – One thing to consider when using a softphone on your Android handset is bandwidth. This might be a problem if you have a cap on your WiFi, but is much more likely to be an issue if you use 3G to make VOIP calls. Here are some quick facts and number to help you decide which codec to use -

G711 – good audio quality – uses around 72MB per hour    * also referred to as ulaw and alaw.

GSM – ok call quality – uses around 25MB per hour

G729 – ok call quality – uses around 21MB per hour    * not included with Asterisk or 3CXPhone

So above are 3 common codec choices with Asterisk. G711 has good quality audio but the trade-off is the amount of bandwidth it uses. If you have a small 3G cap you could quite quickly burn through it. The audio quality with GSM and G729 is OK. It’s perfectly understandable, and whether it’s acceptable or not will depend on the person making/receiving the call and their expectations.

G729 has the lowest bandwidth requirements of the 3 but it is not included with Asterisk by default (and is not available for 3CXPhone at all) as there are license costs to run it. You can add G729 to Asterisk but there is a license cost from Digium (there is an opensource version of G729 for Asterisk but you should be aware of any license restrictions - http://asterisk.hosting.lv/). You will also find that there is a higher cost for Android softphones that can run G729.

I would also recommend something like 3G Watchdog if you need to keep track of how much bandwidth you are using - https://play.google.com/store/apps/details?id=net.rgruet.android.g3watchdog&hl=en

Setup – Installation was straight forward. Just create an extension in FreePBX and then you just need the server IP (or name), extension number and extension secret in 3CXPhone. One thing I did notice was that disabling codecs in 3CXPhone did not seem to work for me. I disabled G711 in 3CXPhone and had G711 and GSM enabled in FreePBX for the extension. The call was still make using G711 and I don’t think that should be the case. If you want to use GSM I would just have GSM enabled for that extension in FreePBX. It would also be worth double checking what codec is used by checking a call in progress!

Making calls - 3CXPhone integrates with your Android contact list so it’s pretty easy to make calls. Just select the contact from the list, or dial the number. One thing to bear in mind is that the format the Asterisk system expects the phone number in may not be the same as your mobile operator. The interface is pretty simple with the ability to put a call on hold or transfer a call. One omission is that there is no call timer shown when a call is in progress. This would have been very useful if you’re trying to monitor bandwidth usage.  (in fact a SIP client with a built in bandwidth monitor would be a great idea …)

Receiving calls - Receiving calls worked OK. Just dial the extension number from another extension, or set the extension as a destination for an Inbound Route in FreePBX. One thing to bear in mind is that with power saving potentially turning off the WiFi or patchy 3G coverage when moving around is that receiving calls using any Android client will likely be a hit and miss affair.

Trouble with SIP - The ability to make inexpensive calls from a mobile phone over 3G has not been lost on the mobile phone carriers! Some actively block SIP connections so you may find that you’re not able to use it. Also, your terms of service may restrict your ability to make VOIP calls. It’s worth checking this before you find that your mobile phone carrier has cut you off!

Final thoughts - Using an Android SIP softphone is a great solution for some uses. If you’re by permanent WiFi that has a good signal and little interference then call quality can be great. If you’re using 3G then quality is certainly going to be hit and miss, probably unusable in a poor 3G signal area, or when being handed off between cell towers. Being able to use the internet to make inexpensive international calls is great. I’ve made calls from New Zealand to the UK using a SIP softphone over 3G and the call quality has been fantastic. It’s not always going to work that way, but when it does it’s amazing!

Another possibility  - If you make regular international calls to the same number then another option is to get a local DDI number and then forward that in FreePBX to the international number. For example I have a UK phone number that forwards to a land line in New Zealand. This way I can call a local UK number (included in my mobile plan minutes) that forwards to regular phone in New Zealand. International phone calls from your cell phone to the other side of the world for 1p per minute!  (plus the cost of the DDI number and any mobile phone plan minutes)

This is a pretty advanced setup but here is a walkthrough for setting up a SysAdminMan VPS as an OpenVPN server and then connecting to it with a TP-LINK router running OpenWRT.

Specifically this router is used - http://www.tp-link.com/en/products/details/?model=TL-WR1043ND. I paid around £40 from Amazon, an absolute bargain for something that will run OpenWRT.

Setting up the router

First you need to flash OpenWRT on to the router. This replaces the original firmware. Here are some instructions for this TP-Link router - http://wiki.openwrt.org/toh/tp-link/tl-wr1043nd?s. I got version 18 of the router and flashed Backfire 10.03.1-rc6 version of OpenWRT.

Next the router was connected via the WAN port on the TP-LINK to my home network. The WAN side of your TP-LINK should be given an IP address from your network DHCP server. It will use this to connect to the Internet.

Now connect a PC to a LAN port using a network cable and you should be given an IP address in the range 192.168.1.0/24

Now make the following changes on the router using a web browser. This will install the OpenVPN software and assign a new IP address to the router. If you use a different subnet you will need to change some settings below to match -

• Assign a password
• Change the LAN network address to 10.10.10.1
• In OpenWRT go to System / Software and click on Update Lists
• Click Available Packages and install OpenVPN
• Click System / Administration and enable SSH on the LAN interface
• Click System / Startup and Enable and Start OpenVPN
• Reboot the router

On the SysAdminMan VPS

For this to work you will need a TUN device assigning to your VPS. Please open a support ticket to request this.

First we’re going to install OpenVPN. This will install from rpmforge, which is set up as standard -

Now, if you’d like you can edit /etc/openvpn/vars and change the settings at the bottom to some sensible defaults. This is not required, but will make creating the certificates easier.

Next we’re going to set up some certificates for OpenVPN. You should run these commands one at a time and answer the questions that are asked -

Now create a file called /etc/openvpn/server.conf with the following settings -

Now we’re going to tell OpenVPN to route traffic to our LAN behind the TP-LINK router -

Now fire up OpenVPN -

Next we need to allow the OpenVPN traffic through the local IPTables firewall -

ON THE TP-LINK ROUTER VIA SSH

Next from your PC we’re going to SSH on to the TP-LINK router at 10.10.10.1 and run the following. This will copy the certificate files we created earlier to the router. X.X.X.X should be replaced with your VPS IP -

Now create a copy of the original OpenVPN config file -

Next we’re going to edit that file and change some settings from the default. These settings start half way down the file under the client configuration settings. Finally reboot the router -

ON THE TP-LINK ROUTER VIA THE WEB GUI

First we’re going to create a new interface that includes the tun interface created by OpenVPN

• Network / Interfaces / Add New Interface
• name – openvpn
• protocol – unmanaged
• interface – tun0

and now we’re going to allow traffic through the TP-LINK firewall to the VPN -

• Network / Firewall / Zones / Add
• Input / Output / Forward = Accept
• Tick lan in Destination and Source zones

and now Reboot the router.

And we’re done!

If you used the settings above then VPS should be accessible on the IP address via the IP 10.20.0.1. You should be able to get to the FreePBX web interface on this address and also use it for your phone configuration.

This is definitely not for the faint hearted as it’s pretty technical and could require some troubleshooting if things don’t work immediately. It’s pretty cool though and should allow for multiple VOIP handsets to be plugged in to the TP-LINK router and connect to the VPS without any worry of NAT problems  (as there’s no NAT happening across the VPN).

I’ve done a few test calls which worked well, but I’ve still got to do some speed tests to see how well the TP-LINK performs.

This post is about Inbound Numbers, and how incoming calls get to your VOIP equipment/server.

What numbers are available?

You can get telephone numbers all over the world. It’s easier to get numbers in some countries than others. You can also get different types of numbers. For example; in the UK you can get geographic numbers, freephone numbers, national numbers and even mobile numbers.

DID or DDI, which is it? 

In the UK we tend to call Inbound numbers DDIs (direct dial-in), the US and other countries call them DIDs (direct inward dialing)

Where do I get an Inbound Number?

There are literally hundreds of call providers that can supply Inbound Numbers. You should choose your provider carefully as it’s more critical to find a reliable number provider, than a reliable outbound call provider. It’s very easy to switch to a new outbound call provider but once you have published your incoming phone number it’s more difficult to move that to a new call provider. Sometimes impossible.

Call providers are referred to as ITSPs (Internet Telephony Service Provider).

How do the calls get to my Voice Server/PBX?

Calls initially go to you Inbound Number provider. They are then delivered to your PBX/Voice Server. This happens very quickly. In the world of Asterisk calls are delivered to your server using either SIP or IAX2. Both of these are VOIP standards. The most common standard is SIP and you will find that most ITSPs can deliver the calls to your Asterisk server using SIP.

How much do they cost?

The cost of numbers vary wildly from country to country. Geographic numbers in the UK are very inexpensive, you can ever get free ones. Numbers in other countries can be more expensive. Normally when you buy a number you will pay a fixed monthly fee. This usually includes a number of concurrent channels, plus a number of included minutes. Once you exceed your monthly minute allocation there is sometimes a per minute cost for additional minutes. If you expect more than a couple of concurrent calls and/or lots of calls to your number then check these 2 restrictions carefully when looking for a number.

For freephone numbers you are likely to get no included minutes and pay a per minute rate for all incoming calls.

Can I get a number in any country?

Not necessarily. For instance, anyone can buy a number in the UK or US. However, to buy a number in France you need to prove that you are a resident by providing a utility bill or similar.

What happens to an inbound call when it gets to my Voice Server?

In FreePBX you control what happens to an incoming call by creating an Inbound Route. It could go to an extensions, voicemail, interactive menu, conference room or one of many other options.

SysAdminMan is now an authorised installer for Humbug Labs. FREE installation on your SysAdminMan Asterisk VPS!

Humbug Telecom Labs provides fraud detection and monitoring for your Asterisk (and other) phone system. If you’ve seen Google Analytics then Humbug Labs works in a similar way, but for VOIP calls. You install a small client on your Asterisk server, which passively sends call detail records to Humbugs servers in ‘the cloud’.

There you can do all sorts of clever monitoring and alerting via the web portal.

This works great with both FreePBX and A2Billing.

Here are the features included -

TECHNICAL - receive notifications when certain technical events occur, events that are strong indicators of pending fraud.

BLACKLIST - be notified of traffic from blacklisted sources:

• User Defined
• Community Blacklist
• Geographical – blacklist entire geographical regions around the globe

TIMESTAMP - be notified when calls originate from your organization at times you deem suspicious.

• User defined time ranges
• Business hours

COST & DURATION - receive alerts when individual calls exceed a specified cost per minute, total cost or duration.

THRESHOLD - receive alerts when daily or hourly costs, durations, or call volume thresholds are exceeded on a total or per country basis.

STATISTICAL - user Specific Statistically Significant Anomalies, receive alerts when we identify traffic that is inconsistend with your historic telecom usage.

There is currently no cost for a Humbug Labs account and SysAdminMan is offering free installation if you have a VPS from SysAdminMan. Please raise a ticket in the support system to request this.

Here are the tips -

• When setting up WiFi profile, be sure to add the SSID by scanning for WifI AP’s.

• When setting up or switching to WiFi profile be sure to let the phone scan for AP’s, and select your SSId from the list.

• Not all 1535 seem to be created equally, mine for instance takes several tries before it finally lists available AP’s.

• When listing fails, it even tells me Wifi Profile not enabled. Ignore that and try again until eventually the AP’s are shown.

• Then pick your SSID and configure the security settings. This way the phone stores the WiFi information differently than by entering your SSID by hand. When entered by hand, on restart the phone will connect fine to WiFi but will not automatically register the SIP account. When you let the phone search and list available WiFi AP’s and you configure the WiFi in this way, the phone will automatically start the registering of the phone after a restart or after switching from LAN to WiFi.

• If the scanning of WiFi fails with “Wireless profile not enabled” , it might help to put the phone further away from your WiFi router. If scanning still fails, you can enter the SSID manually and register manually. When the phone is restarted, there is no guarantee that it will automatically register after restart if the SSID is not set via the scanning option. When set the SSID manually, rebooting the phone via telnet will let the phone register at startup…..

• As the phone I got, has the Turkish branding, I gave a demo at a Turkish restaurant. I set the WiFi security, scanned the WiFi networks, choose the right SSID and it registered fine, even after switching the phone off and on by pulling the power cord out. Back home it could not find any AP’s, entered the SSId manually, gave a reboot in telnet and registration at startup worked fine.

Installation is quick and simple and entering the extension details for my Asterisk server got it registered without any problems.

One of the most interesting things is the variety of codecs that Ekiga supports. The ‘default’ for most people when using Asterisk is G711, an 8kHz codec. This should give the same quiality as a ‘regular’ PSTN telephone call.

Ekiga supports G722, a 16kHz codec, and enabling this, along with a pair of USB headphones (Plantronics Audio 655) it was definitely possible to hear the difference. Audio on calls sounded clearer and fuller. This obviously assumes that the call is in G722 end-to-end. If it goes on to the PSTN audio will be converted to a lower quality there.

Maybe even more interesting is the support for CELT. This is a very high qulity stereo codec. While this is not currently supported in Asterisk, version 10 will offer pass-trhu support for CELT (Asterisk 10 Audio Codecs) and, in the future, hopefully be able to transcode it too.

Ekiga also offers support for Video.

So if you’re looking for a simple, quick, cross-platform SoftPhone then it’s worth checking out Ekiga – ekiga.org

I wanted to run Android 2.3 – Gingerbread – but HTC havn’t released their HTC Sense for the Desire/Gingerbread combination yet, so if you want to run Android 2.3 you can, just without Sense.

I installed CyanogenMod 7.0.2 which is based on Android 2.3.3. The install was easy on my rooted Desire and after having it for a day I don’t think I’ll miss HTC Sense.

One really nice feature is the addition of a SIP stack as standard. I was able to type in the name of my Asterisk server, plus the extension and secret, and start making calls immediately. I’ve used SipDroid before but the integration of SIP in Gingerbread seems much nicer.

I called my Asterisk voicemail over both WiFi and 3G without any problems but, as always with VOIP and WiFi/3G, you are going to want a good signal and not be moving around to get acceptable results.

Here are some screenshots of my Gingerbread install -

Here’s a neat tool that checks your link for latency, packet loss and jitter. Bear in mind that if you’re running the test on a computer/laptop that’s connected via WiFi you are likely to get much worse results that a physically cabled connection. Running VOIP over WiFi is definitely not recommended.

The test requires Java installed on the machine running the test.

The test is provided by Visualware and can be accessed here – http://myspeed.visualware.com/indexvoip.php

http://blogs.digium.com/2011/01/26/pbx-in-the-cloud-or-in-the-closet/

website : http://www.voipusersconference.org/

podcast feed : http://recordings.voipusersconference.org/

twitter : http://twitter.com/voipusers