[Asterisk-Users] Compare to Skype

[mgraves at mstvp.com]

What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required in an
implementation that processed the streams to bridge into the TDM/PSTN
world. It would be great....but don't hold your breath.

For now there are Skype bridges like PSWG and Uplink that interface
Skype to SIP. These are simplistic but sometimes workable.

Does anyone here have experience with Uplink? I tried PSGW and gave up
eventually.

Michael Graves
Sr Product Specialist
Pixel Power Inc
mgraves at pixelpower.com
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262



> -------- Original Message --------
> Subject: Re: [Asterisk-Users] Compare to Skype
> From: Ronald Wiplinger <ronald at elmit.com>
> Date: Sun, April 30, 2006 9:09 am
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> 
> mgraves at mstvp.com wrote:
> >> -------- Original Message --------
> >>
> >> Skype uses iLBC codec, which has great jitter compensation.  IIRC, the
> >> newer SIP channels of * are supposed to have the same capabilities, but
> >> I have not tested.  I really do not like Skype (prefer FWD), but I must
> >> say, over satellite, etc, they provide quality..  All about the codec in
> >> this case..
> >>     
> >
> >
> > Errr...no...this is wrong. 
> >
> > Skype uses ISAC from Global IP Sound. iLBC is something different see
> > http://www.globalipsound.com/solutions/solutions_Codecs.php
> >
> > One of the reasons Skype sounds good is that its a closed system and so
> > can leverage a wideband codec. Instead of the normal 8khz sample rate
> > it uses 16khz. That makes for clearer sound. Since ISAC is a
> > proprietary relative of iLBC its jitter compensation is also very good.
> >
> > My understanding is that Asterisk cannot presently use any wideband
> > codecs as it is hard coded to the 8khz sample rate at its core.
> > Adapting Asterisk to wideband capability has been discussed but will be
> > a huge amount of work. Further, only if you know that the calls will
> > stay wideband end-to-end will the benefits of wideband be apparent.
> > That means no PSTN segments.
> >
> > Michael Graves
> > mgraves at mstvp.com
> >
> >   
> 
> Sadly to say, but users do not care about the why, they only care about 
> the quality! and they simple ask to "fix" it!
> 
> I hope there is soon a solution, otherwise, we have to skip all our 
> effort and just use skype!!!!!
> And I would hate to see that. I just lost 20 US$ to Ebay - the newly 
> parent company of skype, for a not received parcel, but the rules says, 
> below 25 US$ there is no guarantee that you get anything!!!!
> 
> 
> bye
> 
> Ronald Wiplinger
> 
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