[OpenSIPS-Users] SIP and RTP Proxy without local user base

[Stefano Pisani]

Remember that if the audio path is p2p the DMTF tones do not work 
anymore, so you can't transfer a call, etc. You should use SIP INFO if 
supported.
For the audio path give a look to the asterisk options /directmedia/=yes 
and directrtpsetup=yes. You could not need to use openSIPS and rtpproxy.



Il 24/04/2015 17:36, Russell Treleaven ha scritto:
> As an alternative you could try bypass media.
> https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview
>
>
> On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer <roman at dissauer.net 
> <mailto:roman at dissauer.net>> wrote:
>
>     Dear all,
>
>     I’m running a centralized Freeswitch based PBX for use on several
>     sites. All phones register against this Freeswitch instance.
>
>     Now I want to install SIP and RTP Proxies on every site to keep
>     RTP traffic locally on site. Registration should still be done by
>     freeswitch. Can anybody give me a hint if this is possible with
>     opensips and rtpproxy?
>
>     Maybe I can clarify it a bit more:
>     ----------------
>     |  Freeswitch  |
>     ----------------
>            |
>            |    Public Internet
>            |
>            |--------- Phone 3 external
>            |
>            |
>     ------------------
>     | Firewall / NAT |
>     ------------------
>            |
>     ----------------
>     |    Proxy     |   Site 1
>     ----------------
>            |
>            |
>            |--------- Phone 1 internal
>            |--------- Phone 2 internal
>
>     Phone 1 - 3 are all registered at Freeswitch
>     Phone 1 calls Phone 3: SIP and RTP over Freeswitch
>     Phone 1 calls Phone 2: SIP over Freeswitch but RTP over Proxy
>
>     Does this make sense?
>
>     Thanks,
>     Roman
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