ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
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!
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
Ok, I have adapted the default for set_contact_alias and handle_ruri_alias (you meant ruri not uri, right? i cant find handle_uri_alias in the nathelper module)
Now I've tried some new scenarios:
Using JSSIP instead of SIPML5 Calling with a peer connected to Asterisk directly to WebRTC client works fine The inverse scenario (from WebRTC client JSSIP to Asterisk peer) fine
Also tried sip softphone connected directly to Kamailio to WebRTC client and works fine both ways
The problem ocurrs when I call from WebRTC client to WebRTC client Both WebRTC client and SIP softphone (X-Lite) are being used from the same PC
First I tryed the JSSIp functionality "hack_ip_in_contact" http://jssip.net/documentation/0.3.x/api/ua_configuration_parameters/#parame...
(Kamailio log) Aug 8 15:13:12 ieol /usr/local/sbin/kamailio[9632]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Aug 8 15:13:12 ieol /usr/local/sbin/kamailio[9632]: ERROR: <core> [msg_translator.c:1722]: build_req_buf_from_sip_req(): could not create Via header Aug 8 15:13:12 ieol /usr/local/sbin/kamailio[9632]: ERROR: <core> [forward.c:585]: forward_request(): ERROR: forward_request: building failed Aug 8 15:13:12 ieol /usr/local/sbin/kamailio[9632]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
(in Asterisk log) -- Got SIP response 500 "No error (2/SL)" back from 95.85.54.123:5060
But without that property set:
(Kamailio log) Aug 8 15:16:25 ieol /usr/local/sbin/kamailio[9632]: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "g7q0vsqch2ne.invalid" Aug 8 15:16:25 ieol /usr/local/sbin/kamailio[9632]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "g7q0vsqch2ne.invalid" Aug 8 15:16:25 ieol /usr/local/sbin/kamailio[9632]: ERROR: tm [t_fwd.c:1773]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Aug 8 15:16:25 ieol /usr/local/sbin/kamailio[9632]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL)
(in Asterisk) -- Got SIP response 478 "Unresolvable destination (478/SL)" back from 95.85.54.123:5060
I'm a little bit stucked with this :(
Regards, Manuel
2014-08-08 8:14 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
You have to do nat traversal logic for signaling -- see default config file for set_contact_alias() and handle_uri_alias().
Cheers, Daniel
-- Daniel-Constantin Mierlahttp://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
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Daniel-Constantin Mierla -
Manuel Camarg