<div dir="ltr">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)<div><br></div><div>Now I've tried some new scenarios:</div>
<div><br></div>
<div>Using JSSIP instead of SIPML5</div><div>Calling with a peer connected to Asterisk directly to WebRTC client works fine</div><div>The inverse scenario (from WebRTC client JSSIP to Asterisk peer) fine</div><div><br></div>
<div>Also tried sip softphone connected directly to Kamailio to WebRTC client and works fine both ways</div><div><br></div><div>The problem ocurrs when I call from WebRTC client to WebRTC client</div><div>Both WebRTC client and SIP softphone (X-Lite) are being used from the same PC</div>
<div><br></div><div>First I tryed the JSSIp functionality "<span style="color:rgb(0,0,0);font-family:sans-serif;font-size:12px;text-align:right;background-color:rgb(254,254,254)">hack_ip_in_contact"</span></div>
<div><span style="text-align:right;background-color:rgb(254,254,254)"><font color="#000000" face="sans-serif"><span style="font-size:12px"><a href="http://jssip.net/documentation/0.3.x/api/ua_configuration_parameters/#parameter_hack_ip_in_contact">http://jssip.net/documentation/0.3.x/api/ua_configuration_parameters/#parameter_hack_ip_in_contact</a></span></font><br>
</span></div><div><br></div><div>(Kamailio log)</div><div><div>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</div>
<div>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</div><div>Aug 8 15:13:12 ieol /usr/local/sbin/kamailio[9632]: ERROR: <core> [forward.c:585]: forward_request(): ERROR: forward_request: building failed</div>
<div>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)</div><div><br></div><div>(in Asterisk log)</div>
<div><div> -- Got SIP response 500 "No error (2/SL)" back from <a href="http://95.85.54.123:5060">95.85.54.123:5060</a></div></div><div><br></div><div>But without that property set:</div><div><br></div><div>(Kamailio log)</div>
<div><div>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"</div><div>Aug 8 15:16:25 ieol /usr/local/sbin/kamailio[9632]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "g7q0vsqch2ne.invalid"</div>
<div>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</div><div>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)</div>
</div><div><br></div><div>(in Asterisk)</div><div><div> -- Got SIP response 478 "Unresolvable destination (478/SL)" back from <a href="http://95.85.54.123:5060">95.85.54.123:5060</a></div></div><div><br></div>
<div>I'm a little bit stucked with this :(</div></div><div><br></div><div>Regards,</div><div>Manuel</div><div class="gmail_extra"><div><div dir="ltr"><div><br></div></div></div><div class="gmail_quote">2014-08-08 8:14 GMT+02:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
You have to do nat traversal logic for signaling -- see default
config file for set_contact_alias() and handle_uri_alias().<br>
<br>
Cheers,<br>
Daniel<span><font color="#888888">
<pre cols="72">--
Daniel-Constantin Mierla
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