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{April 19, 2007}   forgotten posts

seems I entered some stuff in my kontact journal and forgot to ever post it here. I think I’ll break this up into more than one post… err… if I can remember what else I was going to post about.

over the long weekend, I was messing with voip. I can’t remember what happened when any more, but initially I was expecting to be disappointed. lots of cheap sound cards have this nasty ‘feature’ of recording whatever they’re currently playing – so if you’re trying to have a conversation with someone, their voice gets picked up and echoed back to them. even if you’re wearing headphones. this makes voip unbearable. you really don’t understand how much it breaks your mind until you’ve experienced it.
so, I was happily amazed when I found out my laptop does not have this problem. the sound… just… works. and the mic volume is great, even the builtin mic is usable. I installed skype, and once I got past the segfaults from it hating scim, and restarted it once because it only showed two of my contacts at first, the actually calling part worked fine. :)

so then I figured I’d push my luck by trying to get something non-skype working. I already had a SIP account with wengo from my failed attempts to make SIP work on the home comp, so I installed a handful of SIP proggies and got to testing. it didn’t take too many tries to get the right settings for logging in, and so long as I made sure to suspend artsd the sound worked, but I couldn’t make calls. when I tried the official client, sound wasn’t working at all. eventually I did get sound to work, but there was a lot of crashing, and it was way too late at night for me to be thinking straight. the end result was that the wengo client works fine when it’s not in the mood for constant crashes, while twinkle and kphone are nice and stable but can only call the echo service.
the people at wengo seem to have given up after I gave them logs of all three clients attempting calls. *sigh* guess it’ll be a long time before I can just click a button in kontact to call someone. but hey, at least my lovely little laptop is more than capable.

oh, and zander: I wanted to comment on your post about searching for a laptop, but kdedevelopers.org insists my password is wrong and won’t email me a new one or let me re-register.



Ralesk says:

Hmm, the voice thing, I think it’s more like a setting in ALSA (more precisely the mixer) — it is possible to record the stereo mix, and just the mic (or other input devices). At least, I haven’t had any trouble here doing so.

What tends to be a big disappointment with VoIP is 1) jitter 2) sound quality (terrible codecs or just plain crappy clients, I donno — G.711 should sound decent, but it doesn’t usually!) 3) latency 4) Sound missing due to NAT issues (especially with SIP) — STUN is an okayish workaround for it, forwarding necessary ports and having an UA that respects the ports you set is another.

Had a similar issue with kdedevelopers.org — except I only wanted to register to comment, and I never got an e-mail at all, so I had to go waaaay out of my way and create a drupal.org account :/



Lee says:

Twinkle works great — best linux client I’ve tried so far, by a long way.

That said, SIP in general has some issues which are tricky to get started with. Stick with it though; it’s nice once you get it figured out. Ralesk’s post seems to cover that well (from what I remember).

I’ve just switched to a voip-capable router though, which seems like a very nice way to handle things (along with a static IP).

Personally, I’d avoid proprietary stuff like skype — SIP is way more compatible with the other stuff out there — hardphones, softphones, routers, add-on software, proxies, testing tools, etc. It’s the old closed-vs-open issue all over again.



robbat2 says:

On the software side: I’ve had the most success using Ekiga.

On the hardware side: I picked up a Ipevo handset: http://www.ipevo.com/v2/prod/p111444.htm The buttons don’t work yet. Something in the kernel input layer is losing them, but I’m working on a patch.
Sound quality is superb on the handset, as is the automatic gain for volume control.

On the SIP provider sip: I have a incoming and outgoing POTS service via diamondcard.us – they provide setup instructions for a variety of open-source clients :-). I’ve used both STUN (Ekiga) and SSH tunnels for my SIP connections. Once or twice I’ve had a bad latency on the provider side, but it’s goes away if I wait 10-15 minutes.



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