[Freeswitch-users] RTP traffic on LAn

[Steven Ayre]

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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