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<div class="moz-cite-prefix"><br>
Sorry for the empty message. Pressed the wrong button.<br>
<br>
<br>
I have been wrestling with a pretty generic Asterisk configuration
(version 11.11.0 ) set up with FreePBX.<br>
The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled.<br>
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 <br>
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.<br>
<br>
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.<br>
<br>
This has survived upgrades of everything (firewall, Asterisk
server, workstation)<br>
<br>
This has reduced my Asterisk telephone to an answering machine
with Skype as my way of actually talking to people.<br>
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.<br>
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).<br>
<br>
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.<br>
<br>
Ron<br>
<br>
On 12/03/2015 10:21 AM, Bryant Zimmerman wrote:<br>
</div>
<blockquote cite="mid:a818b026123048dfb27ad1122dac2ce2@zktech.com"
type="cite"><span style="font-family: Arial, Helvetica,
Sans-Serif; font-size: 12px">
<div>Hey all</div>
<div> </div>
<div>We have been working with SIP for years. It has the
potential to be better than Skype. It is really all in the
implementation.</div>
<div>Not all SIP soft clients are equal nor are the networks and
computers they are running on.</div>
<div>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.</div>
<div> </div>
<div>With SIP you get to choose how good your quality is. With
Skype Microsoft does. </div>
<div> </div>
<div>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.</div>
<div> </div>
<div>You can choose to agree or disagree with my statements. I
hope they are useful to some.</div>
<div> </div>
<div>Thanks<br>
<br>
Bryant</div>
<div> </div>
<hr align="center" size="2" width="100%">
<div><span style="font-family: tahoma,arial,sans-serif;
font-size: 10pt;"><b>From</b>: "Ron Wheeler"
<a class="moz-txt-link-rfc2396E" href="mailto:rwheeler@artifact-software.com"><rwheeler@artifact-software.com></a><br>
<b>Sent</b>: Thursday, March 12, 2015 9:40 AM<br>
<b>To</b>: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
<b>Subject</b>: Re: [asterisk-users] switching from SIP to
Skype..or not</span>
<div> </div>
Your characterization may be true but Skype works much better
than SIP<br>
when it comes to sound quality.<br>
<br>
I have SIP softphone with Asterisk server and Skype on the
same<br>
workstation.<br>
Skype just works better over the same network.<br>
<br>
Ron<br>
<br>
On 12/03/2015 9:26 AM, A J Stiles wrote:<br>
> On Thursday 12 Mar 2015, Thufir wrote:<br>
>> I'm testing Asterisk at home, crummy connection.
Skype works fine for<br>
>> me, but every SIP client, even without using
Asterisk, fails to connect.<br>
>> That's ok.<br>
>><br>
>> Is swapping out SIP for Skype a big deal?<br>
> Stay away from Skype! It is a toxic, proprietary product.
The lack of<br>
> interoperability by design is the antithesis of what a
telecommunication<br>
> system should be about -- and the extent to which they
have gone to thwart any<br>
> attempt at interoperability is truly shocking.<br>
><br>
> For connecting two Asterisk installations to each other
over the Internet, IAX<br>
> is better than SIP -- that's what it was designed for.<br>
><br>
<br>
<br>
--<br>
Ron Wheeler<br>
President<br>
Artifact Software Inc<br>
email: <a class="moz-txt-link-abbreviated" href="mailto:rwheeler@artifact-software.com">rwheeler@artifact-software.com</a><br>
skype: ronaldmwheeler<br>
phone: 866-970-2435, ext 102<br>
<br>
<br>
--<br>
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</span>
<br>
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</blockquote>
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<br>
<pre class="moz-signature" cols="72">--
Ron Wheeler
President
Artifact Software Inc
email: <a class="moz-txt-link-abbreviated" href="mailto:rwheeler@artifact-software.com">rwheeler@artifact-software.com</a>
skype: ronaldmwheeler
phone: 866-970-2435, ext 102</pre>
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