> 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.
I Disagree, if you have a teleconf with 2 people you will have to
upstream your voice twice (and also receive twice, but D\L is not an
issue).
the way its being done is a single voice session to a conf server that
mix all the streams into one.
and above everything its a networking nightmare, adjusting each
NAT/Firewall of each client to accept each of the participates in the
conf.
> 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.
I Wouldnt say "just", SIP never comes alone...
Anyways,
I Would just install Asterisk and setup the conf on it using IAX
because its more
NAT/Firewall friendly.
i would use GSM codec (G711 will eat about 80kbit of your bandwidth)
with any IAX softphone out there (or SIP) or if you are willing to put
some money into it, buy a softphone with G729 and on the Asterisk
install the G729 with Educational license.
and if you really want to go further... move this setup to a central
point in the internet like UK or Germany (90% of the time the packets
will go thru those country anyway, when calling outside israel)
/Nitzan.
On 8/22/06, Amos Shapira <[EMAIL PROTECTED]> wrote:
On 21/08/06, Yedidyah Bar-David <[EMAIL PROTECTED]> wrote:
> 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
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