[asterisk-users] switching from SIP to Skype..or not

[Ron Wheeler]

Sorry for the empty message. Pressed the wrong button.


I have been wrestling with a pretty generic Asterisk configuration 
(version 11.11.0 ) set up with FreePBX.
The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled.
I was using Eyebeam and am now trying Jitsi. Jitsi has a number of 
codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and 
telephone-event
The internet connection from the workstation to my internet supplier 
(workstation to firewall/router to speed test server at ISP) tests at 
13MBs incoming 6Mbs outgoing.

The problem has always been great sound from the other telephone and 
choppy sound (dropped sound fragments) from me to the caller with only 
one call going through Asterisk and the network pretty much dedicated to 
the my workstation.

This has survived upgrades of everything (firewall, Asterisk server, 
workstation)

This has reduced my Asterisk telephone to an answering machine with 
Skype as my way of actually talking to people.
This fixes the sound issues and is actually cheaper since I pay a low 
monthly fixed cost for Skype access to all North American telephones.
Skype does not seem to have an problem traversing the same network even 
with two way video active or during multi-party conferences (mix of 
Skype and telephones in the group).

I would like to have a reliable 2 way conversation using Asterisk but 
have not found any suggestions about the source of the problem or how to 
fix it.

Ron

On 12/03/2015 10:21 AM, Bryant Zimmerman wrote:
> Hey all
> We have been working with SIP for years. It has the potential to be 
> better than Skype. It is really all in the implementation.
> Not all SIP soft clients are equal nor are the networks and computers 
> they are running on.
> I will not bash Skype. We have tested it and in most cases choose not 
> to use it. It has it's place and is good for the user that meets it's 
> specific target demographic.  SIP is a sold communications protocol 
> that can communication with codecs of differ audio and video quality 
> levels, and supports industry standard software and hardware endpoints.
> With SIP you get to choose how good your quality is. With Skype 
> Microsoft does.
> It comes down to what do you want to achieve, how much resource do you 
> want to put in to it, and are you committed to a bit more work for a 
> lot more options and better quality, or do you want a quick and easy 
> solution with differing limits. Both solutions have their place.  To 
> me SIP vs Skype is like complaining apples and carrots do you want 
> fruit or veggies you get to choose.
> You can choose to agree or disagree with my statements. I hope they 
> are useful to some.
> Thanks
>
> Bryant
> ------------------------------------------------------------------------
> *From*: "Ron Wheeler" <rwheeler at artifact-software.com>
> *Sent*: Thursday, March 12, 2015 9:40 AM
> *To*: asterisk-users at lists.digium.com
> *Subject*: Re: [asterisk-users] switching from SIP to Skype..or not
> Your characterization may be true but Skype works much better than SIP
> when it comes to sound quality.
>
> I have SIP softphone with Asterisk server and Skype on the same
> workstation.
> Skype just works better over the same network.
>
> Ron
>
> On 12/03/2015 9:26 AM, A J Stiles wrote:
> > On Thursday 12 Mar 2015, Thufir wrote:
> >> I'm testing Asterisk at home, crummy connection. Skype works fine for
> >> me, but every SIP client, even without using Asterisk, fails to 
> connect.
> >> That's ok.
> >>
> >> Is swapping out SIP for Skype a big deal?
> > Stay away from Skype! It is a toxic, proprietary product. The lack of
> > interoperability by design is the antithesis of what a telecommunication
> > system should be about -- and the extent to which they have gone to 
> thwart any
> > attempt at interoperability is truly shocking.
> >
> > For connecting two Asterisk installations to each other over the 
> Internet, IAX
> > is better than SIP -- that's what it was designed for.
> >
>
>
> --
> Ron Wheeler
> President
> Artifact Software Inc
> email: rwheeler at artifact-software.com
> skype: ronaldmwheeler
> phone: 866-970-2435, ext 102
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


-- 
Ron Wheeler
President
Artifact Software Inc
email: rwheeler at artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150313/747fcac5/attachment-0001.html>