will kamailio as proxy behind nat and UACs behind another nat work?
port forward sip and rtp done in the router.
UACs register successfully but no audio.
advertised_address = public_ip advertised_port = sip
both define after the line of listen=public_ip
please advice.
thanks.
What are you using - rtpproxy, mediaproxy etc? There is an experimental patch for advertised_address support in rtpproxy on the internet, but I've never had a chance to try it out. You may want to try spce-v2.2, which comes with mediaproxy-ng supporting the advertised_address out of the box: http://www.sipwise.com/news/announcements/spce-v2_2-release/
On 29.06.2011 16:56, MingHon wrote:
will kamailio as proxy behind nat and UACs behind another nat work?
port forward sip and rtp done in the router.
UACs register successfully but no audio.
advertised_address = public_ip advertised_port = sip
both define after the line of listen=public_ip
please advice.
thanks.
It is possible to use kamailio and rtpproxy behind a NAT. You need to properly craft the config to deal with the fact that kamailio is behind NAT (proper Via, Route, Record-Route headers) and you need to do port forwarding for SIP signalling and rtp.
Regards, Ovidiu Sas
On Wed, Jun 29, 2011 at 9:56 AM, MingHon gminghon@gmail.com wrote:
will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Regards,
MingHon
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
Im using kamailio 3.1.4 as proxy and rtpproxy 1.2.1 in a same server and asterisk 1.6 is at another server both server in the same lan.
im new at kamailio and rtpproxy. do you have to source or example mind to share? i try googling but i cant find any.
thank you very much.
I experimented with it a while ago in my lab and the setup is no longer active. There's no out of the box solution for this, as you need to craft proper Route, Record-Route and Via headers based on the direction of the call. You need to understand how loose routing works in order to set correct headers in outgoing requests.
Regards, Ovidiu Sas
On Wed, Jun 29, 2011 at 12:14 PM, MingHon gminghon@gmail.com wrote:
Hi, Im using kamailio 3.1.4 as proxy and rtpproxy 1.2.1 in a same server and asterisk 1.6 is at another server both server in the same lan. im new at kamailio and rtpproxy. do you have to source or example mind to share? i try googling but i cant find any. thank you very much.
-- Regards,
MingHon
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
Can you give me more detail, could you guide me how to?
thanks you very much..
As I mentioned before, based on direction of the call you may or may not need to force an advertised IP/port (see the advertised_address core parameter and set_advertised_address() core function) Also, you will need to force the same IP in the Route/Record-Route headers (see the rr module). And also you will need to put proper IPs in SDP (see the rtpproxy module). Start with a simple config, make calls, check all the SIP requests, find out what's not properly set and fix the config.
Regards, Ovidiu Sas
On Wed, Jun 29, 2011 at 10:08 PM, MingHon gminghon@gmail.com wrote:
Hi, Can you give me more detail, could you guide me how to?
thanks you very much..
-- Regards,
MingHon
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Andrew Pogrebennyk -
MingHon -
Ovidiu Sas