[Freeswitch-users] RTP traffic on LAn

[Kristian Kielhofner]

This may change soon...

If your endpoints support ICE FS can ignore it with bypass media enabled.

On Fri, Jan 11, 2013 at 7:34 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Unfortunately FS does not (as yet) support it. At least not AFAIK.
> It also would require that both endpoints support it too, and not all do.
>
> In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media.
>
> -Steve
>
>
>
>
>
> On 11 January 2013 11:19, Raimund Sacherer <raimund.sacherer at logitravel.com> wrote:
>>
>> Hi,
>>
>> isn't there allready a protocol that does that? ICE?
>>
>> this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action):
>>
>> --- snip ---
>> ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive Connectivity Establishment). ICE optimizes the media path. This would be the case, for example, when two phones in the same network are calling each other via a long media path through other, external networks. With ICE, the short media path in the same network would be chosen, which will presumably have better quality than the long one. Sometimes this feature will stop you from being able to make calls. When this occurs, switch it off.
>> --- snip ---
>>
>> Link here: http://wiki.snom.com/Settings/user_ice
>>
>> Here seems to be more information, check "Probing Media Path": http://wiki.snom.com/Networking/NAT
>>
>> Best,
>>
>> Raimund
>>
>> ________________________________
>> From: "Chris B. Ware" <chrisbware at yahoo.it>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Viernes, 11 de Enero 2013 10:30:52
>> Subject: [Freeswitch-users] RTP traffic on LAn
>>
>>
>> Hi all,
>>
>> I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local.
>> Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different
>> rooms on the same office) consume a lot of bandwidth.
>>
>> Is there anyone who has found a solution to this problem?
>>
>> I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using
>> private IP (no STUN on phones) and keeping FS out of RTP streaming.
>>
>> Any help will be appreciated.
>>
>> Chris
>>
>>
>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>>
>>
>> --
>> Raimund Sacherer
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>>
>>
>> _________________________________________________________________________
>> 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
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>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>
>
>
> _________________________________________________________________________
> 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
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Kristian Kielhofner