Hi all,
I want to have conference calls over the internet with my family and
friends. I know very little about voip. I used for the last year or so
gnomemeeting, pear-to-pear (opened firewalls on both ends), which was
mostly good enough for two-way calls. Now I want 3+ participants. I also
want some of them to be Windows clients. We tried skype, and it works,
but has several drawbacks (in addition to the things discussed here
about it quite a lot):
1. As far as I understand each client sends its output to each of the
others. I want each client to send once to a server, which will mix all
inputs and send the result to each client.
Why do you care? IMHO direct sending between the clients will utilize the network much better and shorten delays.
2. At least the linux version isn't very stable for me - it hangs after
half an hour of talk or so.
It happened to me ages ago. I think I downgraded from the latest version (to 1.2.18?) and forgot about this. Right now I'm on the 1.3 beta and it seems to work fine too.
BTW - I'm on Debian Etch (just moved from Sarge last week).
3. It has no "mute" button (in the linux version).
Maybe you can mute the mic in the alsamixer?
I also tried using gnomemeeting with openmcu, which seems to do exactly
I've never managed to get gnomemeeting to work for me behind the NAT ADSL modem/router, tried to follow all sorts of port forwarding instructions to no avail. MSN Messenger on Windows 98 works behind NAT, though, and it's currently our only way to do some video-conf (although VERY slow and erratic, but that might be also due to our ancient Windows computer).
what I want, but it didn't work at all. Did anyone manage to use it
with the binaries of Debian (sid or sarge), or compiled from source for
that matter?
It seems that the right protocol to choose is sip and friends, with
which I have very little experience, and for which I could not find
(easily) such a thing.
SIP is just a session-initialtion protocol. It only helps in initiating the call but not the transfer of the voice itself. Try digging voip-info.org for a kick start on this. Once I bought an ATA and forwarded port 5060 and some other TCP port range to it through the firewall I can just pick up the phone and dial VoIP calls, but it won't give me teleconf without some PBX (a-la Asterisk?) in the back.
My current wishlist:
* mix voice and video for several clients
* support both linux and windows, at least one stable good client on each
* server runs on linux
* has (the clients) "mute" button, which also stops sending traffic
(unlike what happens if I simply unplug my mic)
* optionally supports echo-cancellation. If it does, has an option to
disable it. I usually find it very annoying and could not find how to
stop it in either skype or netmeeting.
* optionally support text messages
* optionally support presence
* would rather not need tweaking with firewall settings on any of the
clients, but only (if at all) on the server.
I'd like to hear other's opinions and experiences.
I don't work for Skype and don't like their close-ness but I generally have excellent experience with them talking long distance as well as calling PSTN using SkypeOut. I also (very occasionally) do teleconf between three distant countries. It Just Works(TM).
One thing I did a long time ago to improve Skype quality even further is to open up a TCP port directly to it through the firewall and configure it as what Skype calls "a supernode". It cut down the number of relays from 4 to zero. It supposely allows people to leach on my bandwidth for some traffic but I haven't noticed impact on my link (I'm still working on a way to measure this traffic using iptables, Robert X Cringley says it's significant but it doesn't match my experience so far).
Only drawback is that there is no video support for Linux (yet? I gave up on newer Linux Skype version when they suddenly released 1.3 beta with ALSA support a few weeks ago). I hope one day to install VMware+WindowsXP+Skype.2 in order to get video working.
Gizmo Project (gizmoproject.org) is also mentioned frequently in the VoIP circles. It should support Windows, Linux and video but I haven't tried it for a long while now.
I'd also like to hear about available existing servers that allow free
(gratis) such use. I only tried one of them - fwd, which worked well
with several sip clients (including ekiga and twinkle), but which does
not seem to support conference calls on the server.
FWD, as far as I can tell, is basically a SIP server - it allows clients to find each other, once they are connected (the Session is "Initiated") the audio traffic is supposed to run directly between the clients.
--Amos
