[Freeswitch-users] RTP traffic on LAn
avi at avimarcus.net
Fri Jan 11 13:19:27 MSK 2013
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
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
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.
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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.
> 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.
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> Official FreeSWITCH Sites
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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