Of course this is still far from the holy grail of a presence-centered converged synchronous communications platform, but it is a start and anything is better than today’s stupid mass-market telephony.

Here is my /etc/asterisk/jabber.conf :

[general]
;Auto register users from buddy list
autoregister=yes

;Jabber service label
[asterisk]
type=client
serverhost=jabber.grabeuh.com
username=asterisk@jabber.grabeuh.com/kisangani
secret=my_password
port=5222
usetls=yes
statusmessage="Watching the telephone"

Yes, that’s it : Asterisk is now registered as a Jabber client. There are other ways to do it using external modules, but this one is the simplest – and since it is now part of the main trunk it is probably the most stable. In particular you should be careful with class.jabber.php which is not maintained anymore and in the process of being replaced with the much more modern XMPPHP.

Now let’s declare a macro for taking full advantage of that. This one takes a look at the user’s presence and routes the call accordingly : if he is online or chatty the call goes to his desk phone – otherwise it goes his mobile phone.

[macro-reach_user]
; ${ARG3} is a destination such as SIP/whatever
exten => s,1,jabberstatus(asterisk,${ARG2},STATUS)
; presence in will be 1-6.
; In order : Online, Chatty, Away, XAway, DND, Offline
; If not in roster variable will = 7
exten => s,2,gotoif($[$[${STATUS}]<3]?available:unavailable)
; GotoIf(condition?label_if_true:label_if_false)
exten => s,3(available),jabbersend(asterisk,${ARG2},"Call from
 ${CALLERID(name)} at number${CALLERID(num)} on
 ${STRFTIME(,GMT-1,%A %B %d %G at %l:%M:%S %p)}")
exten => s,4,Dial(${ARG1})
exten => s,5(unavailable),Dial(${ARG3})

Since we have declared a macro, we have to call it in the context of our choice and assign the relevant values to the macro’s variables :

[whatever_context]
; ${ARG1} is the destination when at desk
; ${ARG2} is a jabber address used at desk
; ${ARG3} is the destination when not at desk
exten => 05600047590,1,Macro(reach_user,SIP/jml-desk,
 jim@jabber.grabeuh.com, SIP/freephonie-out/0666758747);
; repeat last line for each user

That’s all folks ! That is all it takes to have your calls routed to the right phone according to your presence status. It is really that easy.

Why no service provider is offering that is beyond me. The big ones are all waiting for the IMS systems they are going to deploy with a five years roadmap. But if you want the future right now there is no need to wait : all the technology is here today waiting for you to play with it ! And of course it is 100% Free software…

Some people have even concluded the resulting quality of service is good enough for voice traffic. But even with the Debian Wondershaper ruling my ADSL link I noticed that SIP and IAX still suffer too much packet loss with the saturating traffic occupying the background. I needed better traffic control.

As usual, being a late adopter I am not the only one to have hit that obstacle, and solutions have already been put forth. After rummaging through various mutations, I found Robert Koch’s version of the Wondershaper for the Asus WL-xxx documented on the Wondershaper package page of the WL-500G wiki to be quite promising. Compared to the standard version it prioritizes VOIP traffic by source port for idiot proof configuration, but also by type of service which is much more flexible and can be used thanks to Asterisk being capable of correctly setting TOS fields. As a bonus, using TOS also makes this version of the script capable of distinction between console interactive SSH traffic and bulk SCP traffic using the same protocol and port. And to top it all, it is based on the better hierarchical token bucket (HTB) discipline which is standard since Linux 2.4.20 while the Debian Wondershaper version uses the more based queuing which used to be the more widespread one.

The first shortcoming I found is that it prioritizes SIP and RTP but not IAX and others which I’ll have to add using the SIP stanzas as templates. The other is that taking lists of low priority ports as arguments could make the command line messy and configuration puzzling for the inexperienced user, so I prefer to have this configuration item as a documented variable allocation inside the script. But those are trifles compared to the new VOIP support, enhanced SSH discrimination and overall upgrade.

Hacking on the script I couldn’t resist reorganizing a few things. I originally intended to provide a diff, but that would be pointless since I ended up touching most of the lines. Also be warned that I do not understand why putting ‘prio 1′ everywhere makes the script work whereas other ‘prio’ values at various places made traffic end up in the wrong class and did not make sense at all. In effect, I think that by putting ‘prio 1′ everywhere I just eschewed the use of priority bands inside the classes, which is just fine with me for the intended use. But this show that my tc fluency is still limited and that there are therefore surely ways to enhance this script. I’ll also welcome feedback – whether it works for you or not.

Anyway – it works ! I had a few VOIP conversations across an IAX trunk with lots of background traffic on the uplink and no perceptible effects on voice quality. Life is good. Now that I have removed the last obstacle to taking full advantage of VOIP at home. Soon all my traffic will be routed through Asterisk and there shall be no more RJ11 nor their French T-sockets alter ego in my home.

Here is my modified wondershaper script in all its glory – contrary to Robert Koch’s version it is a drop-in replacement for Debian’s package. Inheriting from the original Wondershaper it is licensed under the GPL so enjoy, modify and share !