VoIP Mailing List Archives

[hawat.thufir at gmail.com]

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?


Heh, well, I guess it's dead:

http://www.digium.com/en/products/software/skype-for-asterisk




If I have a really bad connection, can I "downgrade" SIP somehow? I
don't really need to use to make voice calls. Or, more specifically,
quality, echo, distortion aren't relevant. Just SIP to SIP "hello".


When I connect to any SIP provider, ekiga, etc, without using Asterisk, I
get "too many hops" errors. While I have another computer on the LAN I
can connect to, it's not quite the same.

Any thoughts?



thanks,

Thufir


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[hawat.thufir at gmail.com]

On Thu, 12 Mar 2015 12:52:46 +0000, Thufir wrote:

[asterisk_list at earth...]

On Thursday 12 Mar 2015, Thufir wrote:

[rwheeler at artifact-s...]

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:

[andres at telesip.net]

On 3/12/15 9:39 AM, Ron Wheeler wrote:

[BryantZ at zktech.com]

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@artifact-software.com> Sent: Thursday, March 12, 2015 9:40 AM To: asterisk-users@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@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

[EWieling at nyigc.com]

Which wideband codec did you use when testing SIP?

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, March 12, 2015 9:39 AM
To: asterisk-users@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


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_____________________________________________________________________
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[tzafrir.cohen at xorco...]

On Thu, Mar 12, 2015 at 10:04:08AM -0400, Andres wrote:

[hawat.thufir at gmail.com]

On Thu, 12 Mar 2015 10:04:08 -0400, Andres wrote:

[rwheeler at artifact-s...]

On 12/03/2015 10:21 AM, Bryant Zimmerman wrote:

[rwheeler at artifact-s...]

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:

[hawat.thufir at gmail.com]

On Fri, 13 Mar 2015 11:46:21 -0400, Ron Wheeler wrote: