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[Freeswitch-users] [Remote SIP client] Couple of questions


 
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codecomplete at free.fr
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PostPosted: Thu Apr 16, 2009 9:53 am    Post subject: [Freeswitch-users] [Remote SIP client] Couple of questions Reply with quote

mercutioviz wrote:
Quote:
Just following up... did you get these questions ironed out?

Not yet, but I do need to have a clear understanding about how to set things
up when NAT is involved, especially when remote SIP users are also behind a
NAT router, and especially if they can't make any change to it (eg. staying
in a hotel, or don't have the skills required to open up ports).

Thank you.
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msc at freeswitch.org
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PostPosted: Thu Apr 16, 2009 6:13 pm    Post subject: [Freeswitch-users] [Remote SIP client] Couple of questions Reply with quote

definitely stop by IRC and talk real-time with others who've dealt with this kind of thing.
-MC

On Thu, Apr 16, 2009 at 7:44 AM, Fred-145 <codecomplete@free.fr (codecomplete@free.fr)> wrote:
Quote:


mercutioviz wrote:
Quote:
Just following up... did you get these questions ironed out?


Not yet, but I do need to have a clear understanding about how to set things
up when NAT is involved, especially when remote SIP users are also behind a
NAT router, and especially if they can't make any change to it (eg. staying
in a hotel, or don't have the skills required to open up ports).

Thank you.
--
View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p23079426.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



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codecomplete at free.fr
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PostPosted: Fri Apr 17, 2009 3:27 am    Post subject: [Freeswitch-users] [Remote SIP client] Couple of questions Reply with quote

mercutioviz wrote:
Quote:

definitely stop by IRC and talk real-time with others who've dealt with
this
kind of thing.

Thanks for the tip, but since this firewalls are a frequent issue with SIP,
I think it'd be very useful if someone who has good experience with this
wrote an article in the FS wiki for everyone to benefit.

For instance, at this point, I don't know how data (SIP and RTP) flows
between a remote SIP phone behind a NAT router, and a VoIP gateway located
in the same LAN as a Freeswitch server both behind a NAT router. Do RTP
packets go directly from the gateway to the remote SIP phone, or do they go
through the FS server?

That sort of thing Smile
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