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 <[email protected]
<mailto:[email protected]>> 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
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
