[Asterisk-Users] why even use SIP

Roman Zhovtulya roman at fh-offenburg.de
Tue Mar 22 14:01:32 MST 2005


Hello Scott,
Thanks a lot for the info. I’ll need to do a bit more testing with 3 to
5 simultaneous calls to see if there is any problem.

I don’t really like the hardphones, because you don’t have all the
flexibility offered by the sofphone solution (click-to-call,
paste-n-call, hands-free talking with a headset, etc). Moreover,
headsets are much cheaper than the hardphones.


Actually I’m using IAX to hook to VoipJet.com and there seems to be no
NAT between my server and that of sipsnip.de.

Maybe your Asterisk version was pretty old? Perhaps the problems are
solved in the new releases (I’m now using a month-old CVS version).

Roman





> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Scott Bussinger
> Sent: Montag, 21. März 2005 22:32
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] why even use SIP
> 
> 
> > I'm testing a softphone-only setup (SJPhone with Plantronics 80 
> > Headsets plugged into Soundcard) with around 40 users for that are 
> > linked over LAN in an organization of around 300 people and 
> never had 
> > any of the problems you described (the test is going for 
> over a month 
> > now).
> 
> I'm glad it worked for you. In my case it generally worked 
> pretty well for me as an individual, but as soon as we ramped 
> it up for real use by more than 1 or 2 people it pretty much 
> fell apart. X-Lite had the feature set we needed, but voice 
> quality was terrible and was full of clicks. All of the IAX 
> softphones based on the iaxclient library had the same 
> problem with too long a delay in them. Most of the softphones 
> were not useful to use because they were frankly too hard to 
> use for average users. And on it went.
> 
> > I'm using iLBC codec and the Asterisk is running on a PIII 
> PC with 256 
> > MB RAM :-)
> 
> I tried ulaw, gsm, and speex codecs and the results were 
> pretty similar for each. I ended up using ulaw internally and 
> gsm to connect to the external provider (to cut down 
> bandwidth requirements).
> 
> > Outgoing calls to landline via VoipJet and sipsnip.de, 
> incoming from 
> > FWD (landline number provided by ipkall.com for free).
> 
> I use FWD for my phone at home to connect to the office. I 
> haven't had the energy to tackle the SIP over NAT issues yet. :)
> 
> > We are pretty happy with this setup so far. Do you think 
> there might 
> > be problems later?
> 
> If it's working for you, great! Our problems were noticable 
> as soon as we had 4 or 5 people on the phones at the same 
> time so as long as you test it carefully under realistic use 
> you should be fine. I never could figure out why my systems 
> would never work well, but it just wasn't worth spending any 
> more time on (cheaper to just buy the hardphones which didn't 
> have the problem).
> 
> Be seeing you.
> 
> 
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