[SR-Users] ERROR: <core> [resolve.c:1733]: sip_hostport2su():

[Manuel Camarg]

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!

[Daniel-Constantin Mierla]

You have to do nat traversal logic for signaling -- see default config
file for set_contact_alias() and handle_uri_alias().

Cheers,
Daniel

Post by Manuel Camarg
WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5
sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
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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[Manuel Camarg]

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/#parameter_hack_ip_in_contact

(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

Post by Daniel-Constantin Mierla
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
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