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!
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