[Freeswitch-users] RTP traffic on LAn

[Avi Marcus]

Seems there are two options... if the phones have a static location, you
can set a user variable for location. If they match on both phones, then
it's internal.
Or, check the contact IP for both lines -- if both are the same IP, it's
the same network.
sofia_contact gets you the entire contact string... I don't see how to get
the ip directly.
You can do "sofia status profile internal reg 1000" but the you have to
parse it.

One caveat: if there's multiple registrations turned on, you can have a
user's desk phone and a smartphone sip client on different IPs. So you'd
have to calculate if it's safe to add the bypass on each   leg separately.

Does uuid_simplify help in this case? I see no documentation on how it
worked and finding it in the source code wasn't immediately helpful.

-Avi Marcus

1-718-989-9485 (USA)
1-866-202-5850 (USA & Canada Toll Free)
02-372-1570 (Israel)
020-3298-2875 (UK)


On Fri, Jan 11, 2013 at 12:07 PM, Steven Ayre <steveayre at gmail.com> wrote:

> All you need to do is either set bypass_media=true,
> or bypass_media_after_bridge=true.
>
> The hard part is knowing when to do so. You'll need to compare the
> incoming and outgoing IP addresses to see if they're on the same network,
> and there's no way really to do so without knowing those networks in
> advance. The phones aren't able to tell you. If you set bypass_media and
> the phones can't route directly to each other then all that happens is
> nothing gets heard, it doesn't cause any error or fail the call.
>
> I was thinking on using SIP INFO for sending DTMF (in case customer have
>> to use some phone features activated via phone)
>
>
> bypass_media_after_bridge=true will let FS collect RTP including DTMF
> during a IVR menu, then bridge to an endpoint and only have media going
> directly between caller+callee.
>
>  -Steve
>
>
>
>
> On 11 January 2013 09:30, Chris B. Ware <chrisbware at yahoo.it> wrote:
>
>> 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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>
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>
> 
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