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→ posts→ VoIP in Neo Freerunner with Qtmoko and Linphone

I keep using a Neo Freerunner with Qtmoko in a Debian Squeeze (+ some Wheezy) chroot as my daily phone. It also works well for SMS, GPS, MUA, IM, IRC and some Web browsing with Aurora (using both Wifi and GPRS).

Last weekend I managed to play a little with VoIP possibilities in Qtmoko and ended with a functional SIP client in my hands. I want to share some points which may be useful for other brave freerunners:

  • Qtmoko has a native VoIP client but it seems to be broken. Its sipagent uses port 5060. I couldn't find a way to disable it at qpe startup, so I had to setup my client to use another port.
  • Choosing the application: Ekiga takes 93.4MB of additional disk space. Qutecom takes 212MB. Too much for a 256MB rootfs. Twinkle is ok on this point, it uses about 40MB of additional space, but it fails in the capture (mic) quality. I've tried many setups in my voip-handset state file with no success. Perhaps it's something related to the codecs and/or default parameters. I didn't go deeper on this and gave Linphone a try. This package also uses about 40MB of additional space and finally made my day.
  • Linphone gui is called by linphone-3, which has a character not supported by /opt/qtmoko/etc/qx/profiles.conf and /opt/qtmoko/etc/qx/favourites.conf. So if you want to have some setup here you need to create a link/wrapper/whatever without this dash.
  • You can edit your preferences directly in your ~/.linphonerc . Remember to change the sip_port entry to any other than 5060.
  • I use a basic VoIP service from diamondcard.us. It's cheap and works properly. This company gives financial support to Ekiga and Twinkle projects.
  • I was not able to have default Freerunner headset working on an acceptable quality. Help will be very welcome.
  • Here is my current config:

UPDATE: I didn't make clear that I get a very nice audio quality using internal mic/speaker with the above voip-handset state file. The issue is only when using the headset, which demands understanding and setting up a proper state file.

Enjoy some shots:

Telepathy would be better
Much better would be if QtMoko used Telepathy, then you could call folks on SIP using the same UI as on GSM.
Comment by Anonymous — Mon Aug 29 16:40:47 2011
switching between gsmhandset and voipheadeset / h264
does the switching between the two audiopaths happen automatically or do you have to have a command line call before and after the call. also did you give it a try to enable video from coming in from the external using the h264 codecs one can download from the linphone side?
Comment by Anonymous — Tue Aug 30 04:36:33 2011
comment 3

I switch to the voip state file before calling linphone in the profiles file:

init=alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore

I don't need to switch back to gsm as it seems to be done automatically by the default caller.

No, I didn't try video.

Comment by tiagovaz [wordpress.com] — Tue Aug 30 12:20:46 2011
do qtmoko's volume settings influence the voiphandsettings?

One thing I had always wondered was if the voiphandset or also the gsmhandset settings are altered if one edits the default volume of the mic and the speaker in the qt settings dialog. Do you have any experience with this or know the answer?

regards

robin

Comment by Anonymous — Tue Aug 30 12:34:01 2011
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