You have to do nat traversal logic for signaling -- see default config
file for set_contact_alias() and handle_uri_alias().
Cheers,
Daniel
On 07/08/14 21:16, Manuel Camarg wrote:
I have this config:
WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5
When placing a communication between two sipml5 points I get this errors:
ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR:
sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
ERROR: <core> [forward.c:532]: forward_request(): ERROR:
forward_request: bad host name df7jal23ls0d.invalid, dropping packet
ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error
used: Unresolvable destination (478/SL)
If I place a comm between WebRTC sipml5 -> Kamailio -> Asterisk ->
Softphone (connected to Asterisk)
Everything works fine
I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues
SIP INVITE signalling Ringing 180 message and 200 OK gets done fine
Any ideas where this issue might come from? Looks something related
with media traffic (RTP).
Kind regards!
_______________________________________________
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
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
_______________________________________________
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
