[Freeswitch-users] External Softphone vs. Internal Question

[Michael Collins]

The best way to learn more about this is Tony's "117" post:
http://www.freeswitch.org/node/117

Also, check out Tony's "History of FreeSWITCH" in the FS book. (Note: we
are nearly done with the 2e of the book, so don't buy the old one unless
Packt gives you assurances that you can get the new one as well.)

-MC


On Wed, Apr 24, 2013 at 1:13 AM, Jeff Bernhardt <jeff at askcornerstone.net>wrote:

>  Thanks for taking the time to answer. I know it gets busy around here
> with all sorts of stuff that frankly is over my head! It’s kind of nice
> that way, though… keeps some of the mystery and excitement alive for what’s
> possible.****
>
> ** **
>
> Yeah, I didn’t mean it like “Asterisk can do this so what the hell is
> wrong with Freeswitch?” Was just wondering why, so thanks for the clear
> explanation. ****
>
> ** **
>
> I actually didn’t know Asterisk had so much goofiness. Can you (or anyone
> else) give any examples of its goofiness? We’re relatively light PBX users
> in general (just the basics for clients with no more than 150 phones, some
> with only 5 phones!), so we might not have come across any of them. ****
>
> ** **
>
> Jeff Bernhardt****
>
> Systems Administrator****
>
> Cornerstone Consulting****
>
> 808.440.2900****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, April 23, 2013 7:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] External Softphone vs. Internal Question
> ****
>
> ** **
>
> Hi Jeff,****
>
> The short answer is that you are not forced to create a separate profile
> for internal vs. external phones. However, FreeSWITCH gives you this
> freedom whereas Asterisk does not. You *could* try to cram everything into
> port 5060, but there's no compelling reason to do so. A lot of VoIPers are
> accustomed to using 5060 and only 5060, come what may. FreeSWITCHers
> generally view that as a limitation, not a feature.
>
> By having multiple SIP profiles - quite literally multiple SIP UAs - you
> have more freedom and flexibility to handle goofy scenarios like dealing
> with broken NAT devices. You can put all your broken stuff on a different
> profile and not have to worry that setting a particular option to fix one
> device will break another device. ****
>
> Oh, and keep in mind that "just because Asterisk can do it" doesn't mean
> that Asterisk does it correctly. There are a lot of devices out there that
> "work" but only because they all choose to be synchronized in their
> goofiness. Reams have been written about how FS does not pander to broken
> devices so I won't belabor the point here. Just know this: FS is relatively
> strict in adhering to specs and standards, so if something works with
> Asterisk (or whatever VoIP software) but not with FS then most likely it's
> a matter of figuring out how to tell FS to emulate the brokenness for the
> sake of interoperability.****
>
> Hope this helps. Let us know how your setup is coming along. Be sure to
> use pastebin.freeswitch.org to share any configurations or logs with us.
>
> Thanks,****
>
> -MC****
>
> ** **
>
> On Sat, Apr 20, 2013 at 2:50 AM, Jeff Bernhardt <jeff at askcornerstone.net>
> wrote:****
>
> Hi. I have the following basic setup questions: ****
>
> ** **
>
> When using a softphone (Bria on iPhone) from external (on a different
> external ip address), I could register but no audio would be passed either
> way for any calls. I saw that I should set ext-rtp-ip in the internal sip
> profile to my external ip address (it was on auto-nat, which apparently
> wasn't working) in this wiki http://wiki.freeswitch.org/wiki/NAT_Traversal
> ****
>
> ** **
>
> That didn't work, so I also set my ext-sip-ip to my public ip. After that,
> I could pass audio.****
>
> ** **
>
> However, if I register the phone internally instead and call for instance
> the IVR test line, the call drops after 30 seconds.****
>
> ** **
>
> So it's either no audio when registered externally or 30 second calls when
> registered internally.****
>
> ** **
>
> I found this wiki:
> http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios****
>
> I fall into either scenario 2 or 3, and for both, it says to create a
> dedicated profile for external registrations and put them on port 5090,
> which works. However, is there no other way to solve this problem that
> doesn't require the use of an additional profile on port 5090 but also
> doesn't cut off internally registered calls after 30 seconds? On Asterisk,
> there's no need to open a second port to register external phones. What's
> different about Freeswitch?****
>
> ** **
>
> Also, I don't know what role these play, but I also get these errors:****
>
> [WARNING] switch_core_media.c:1282 Asynchronous PTIME not supported,
> changing our end from 0 to 20****
>
> at seemingly random times****
>
> ...and....****
>
> [INFO] switch_nat.c:590 NAT port mapping disabled****
>
> when I make a call from internally or externally registered softphone to
> external number.****
>
> ** **
>
> Thank you.****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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