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@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@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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