Hi,

I am very new to VoIP, but ulimately I would like to voice chat with my brother. We both have VERY exceptionally slow dialup connections- max bandwidth is ~7 kb/s; when I download it usually slips to 1.7 kb/s. But I know that there are high-compression codecs out there, and even variations on SIP and h.323. I use Redhat 9 and my brother uses Windows 2000. What seems to be the best solution? I tried Linphone (too buggy and unstable), Kphone (absolutely will not compile), Gnophone (doesn't like my sound system, only uses GSM). Gnomemeeting comes close, but I don't _think_ it supports the really low bitrate codec that Netmeeting uses, which is supposed to support 14.4K dialup. Correct me if I'm wrong. But are there any other better ideas you have? How about some of the chat servers that internet gamers use? Teamspeak is supposed to be extremely low bitrate. I just need something with decent voice quality, that is fairly easy for me to set up, and interoperable with Win2K clients. If this question is out of the scope of this forum, can you suggest another? Thanks a lot!!

Let me put it this way... Does anybody use teamspeak? Good, bad? Thanks!

Do a search on 'VoIP' here at linuxquesitons and you might find something.

Has anyone tried the Voip Manager on Linux.

http://www.inlab.csp.it/voip/

let me know how to configure this ... your help is appreciated.

Thanks
Kiran

teamspeak :
independant server ( but no P2P ) very good voice quality, multi users, win interoperability , no video

gnomemeeting:
P2P , video ( multi users throuh openMCU but difficult to make it work )
win interoperabilty with netmeeting

gphone : simple but no directory, no video, no conference

linphone : never succeed to make it working

gnophone : same thing

gyach: no voice success, only seen windows users webcams

gspeakfreely: some problems with voice , no video ( it s a long time i have not tested )

Hmm, just installed kphone seems to be simple, just only one barrier, when you want to set your
identity and register there is asked things like:
User Part of SIP URL:
Host Part of SIP URL:
Outbond proxy , q-value between 0.0-1.0 etc etc, unfortunetely there is no explanation or whatever from that kind or suggestion what there should be filled in (for example what is SIP???) I think that is forgotten
that most potential users off this programs have no idea how to fill in this kind off questions.

Has anyone tried the Voip Manager on Linux.

http://www.inlab.csp.it/voip/

let me know how to configure this ... your help is appreciated.

Thanks
Kiran