00:01 <bencoh> zomg, ofono website has been merged in intel's 01.org

00:01 <bencoh> (or is that o1.org)

05:06 <Oksana> Quiet... I am interested more not in CyanogenMod itself, but in LunaOS

07:33 <freemangordon> jonwil: BTW gcc 4.2.1 is totally unusable for that REed NEON stuff. I finished one of the functions in xprot (still not pushed) and it produces code, which I guess will be slower compared to stock by a factor of 2

07:35 <freemangordon> on the other hand, maybe having -voce done won;t be that hard - there is lots of code in nemo/meego/harm

07:35 <freemangordon> foss code that is

08:17 <jonwil> All that stuff labeled "aep" is very much closed and very complex from what I can see. aep module from harm is 171k or so

08:17 <jonwil> arm binary that is

08:18 <jonwil> x86 is only 18k meaning a lot of that 171k is likely armcc stuff

12:00 <gurki> hum

12:01 <gurki> sb highlighted me, but its too long ago for my log to show the hl msg

12:01 <gurki> :(

12:03 <kolp> <jake42> gurki: so... are you still interessted in working on the design of the spacerframe?

12:03 <kolp> that one?

12:04 <infobot> http://irclog.whitequark.org/neo900/2014-10-30#10679354;

12:50 <gurki> yup

12:51 <gurki> thx :)

14:18 <enyc> bencoh: does SIP (preferably with zrtp) over vpn over ipv6 tunnel work though? rather htan ''gsm telephony''...

14:41 <bencoh> ?

14:42 <bencoh> enyc: replicant/gta04 ? no idea

14:59 <enyc> bencoh: err on neo900 ... anyway .. it was something yo uwere talking about before ;p

15:06 <bencoh> oh ... then I dont see why it wouldnt :)

15:23 <enyc> well the n900 maemo client appraently breaks with sip over vpn

15:23 <enyc> hrrm

15:24 <bencoh> how comes ?

15:24 <enyc> no idea why...

15:25 <bencoh> probably a software issue

15:25 <enyc> indeed =)

15:25 <enyc> https://bugs.maemo.org/show_bug.cgi?id=1860

15:25 <bencoh> (as opposed to hardware ;)

15:25 <enyc> indeed =)

15:26 <enyc> but if srtp sip or sip over vpn, actually works well.. then i care rather less about 'telephony' on neo900

15:26 <bencoh> uh .. ok, seems ugly

15:43 <DocScrutinizer05> sounds like lack of proper SIP config on both ends

15:44 <DocScrutinizer05> use STUN to solve "IP of sender" issue. the problem prolly is that SIP needs to know about own public IP to tell far end where to send RTP data to

15:51 <DocScrutinizer05> actually using asterisk on own box just for SIP is pretty much a defect concept. asterisk is no SIP server but a B2BUA

15:51 <bencoh> :)

15:53 <DocScrutinizer05> ~wiki b2bua

15:53 <infobot> At http://en.wikipedia.org/wiki/B2BUA (URL), Wikipedia explains: "A 'back-to-back user agent' ('B2BUA') is a logical network element in Session Initiation Protocol (SIP) applications.RFC 3261, "SIP: Session Initiation Protocol", IETF, The Internet Society (2002) SIP is a signaling protocol to manage multimedia Voice over Internet Protocol (VoIP) telephone calls. A back-to-back user agent operates between both end points of a phone call or ...

15:56 <DocScrutinizer05> also see http://en.wikipedia.org/wiki/Peer-to-peer_SIP

15:57 <DocScrutinizer05> which telepathy soesn't really support too well

15:57 <DocScrutinizer05> doesn't*

16:12 <DocScrutinizer05> also see http://sipforum.org/pipermail/discussion/2010-November/007989.html

16:13 <DocScrutinizer05> note that fremantle dialer / telepathy supports ar "allow loose routing" checkbox

16:17 <DocScrutinizer05> http://www.iptel.org/sip/intro/scenarios/rr/strict_vs_loose

16:18 <DocScrutinizer05> http://en.wikipedia.org/wiki/STUN for the proper way of UA to detect own IP

16:20 <DocScrutinizer05> when your SIP server / registrar is in LAN (VPN + local asterisk) then you probably need asterisk to support STUN so the UA can detect own "public" (LAN-wide) IP address

16:21 <DocScrutinizer05> regardless of VPN not really involving a NAT

16:25 <DocScrutinizer05> you won't want to have session control via VPN but RTP via public netwrok interface, right?

16:26 <DocScrutinizer05> o/

16:26 <DocScrutinizer05> n8

16:27 <kolp> shouldn't everything go over the vpn?

17:38 <jake42> gurki: ping

17:55 <enyc> DocScrutinizer05: hrrm no, and you usually can configure Asterisk so that RTP is relayed though asterisk

17:56 <bencoh> you can even do that per-client iirc

17:57 <bencoh> (noreinvite/ canreinvite/reinvite option, I havent played with that for a while)

19:39 <enyc> I do wonder why asterinsk rather than freeswichch used often, but thats another story =)

22:21 <DocScrutinizer05> yes, but relaying via asterisk only makes sense for a local exchange with multiple extensions, then your asterisc would conect to the world as a single UA (e.g. to sipgate.de) or even via landline, and your SIP extensions/UAs register to asterisk. Usually that's not the ideal setup, it's kind of overkill for "normal" SIP scenario where your UAs register directly to e.g. sipgate.de

22:22 <DocScrutinizer05> nevertheless firtzbox routers implement exactly such scenario, and my N9 works great with fritzbox

22:22 <DocScrutinizer05> dunno which B2BUA fritzbox implements

22:24 <DocScrutinizer05> I dunno how a VPN would spoil this scenario, as long as routing between fritzbox B2BUA and client works

22:32 <DocScrutinizer05> n8