Hi Daniel,
thanks for you e-mail... i will try that approach.
in the main time i exploited the "received" parameter in route header sent by freeswitch and with a bit of manipulation i managed to rout the call back to WS client :=)
Im will definitely try to enable r2 and come back with the results.
--- snip ---
route[SIPOUT] { if (!uri==myself) {
xlog("L_INFO", "ROUTE HEADER value: $hdr(Route)\n");
$var(rcv_regex)= $(hdr(Route){re.subst,/.*received=(.*)/\1/});
$var(ip)= $(var(rcv_regex){s.select,0,~}); $var(port)= $(var(rcv_regex){s.select,1,~}); $var(proto_number)= $(var(rcv_regex){s.select,2,~});
if ($var(proto_number) == "1") { $var(trans) = "UDP";
} else if ($var(proto_number) == "6" ) { $var(trans) = "ws";
} else { # not interested for now $var(trans) = "TCP"; }
$var(request_route_regex) = $var(ip) + ":" + $var(port) + ";transport=" +$var(trans);
xlog("L_INFO", "ROUTE HEADER - rcv_regex: $var(rcv_regex)\n"); xlog("L_INFO", "ROUTE HEADER - request route : $var(request_route_regex)\n");
append_hf("P-hint: outbound\r\n"); $ru="sip:" + $rU + "@"+$var(request_route_regex); xlog("SCRIPT: Request uri = $ru\n"); route(RELAY); } }
--- snip ---
On 13 February 2018 at 11:48, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
you have an explicit set of $fs, which is setting the send socket. You have to do then when receiving from WS, not when going to WS.
In 5.1, you can try to enable:
html#path.p.enable_r2
and then let kamailio do all about send socket (remove the assignments to $fs).
Cheers, Daniel
On 09.02.18 22:57, Tihomir Culjaga wrote:
Hi,
Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929. 0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77; branch=z9hG4bK6931435 Max-Forwards: 69 To: sip:1001@192.168.50.60 From: sip:1001@192.168.50.60;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri= "sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: sip:1001@192.0.2.110;transport=ws;reg-id=1;+sip. instance="urn:uuid:587f4694-7e55-46dc-a858-579a82556a20";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: sip:192.168.50.60;lr;received=192.168.200.77~63296~6
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929. 0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77; branch=z9hG4bK6931435 From: sip:1001@192.168.50.60;tag=vq30modpgd To: sip:1001@192.168.50.60;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: sip:1001@192.0.2.110;transport=ws;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~ b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: sip:192.168.50.60;lr;received=192.168.200.77~63296~6 Content-Length: 0
Once, registered, im trying to originate a call to the registered endpoint from freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: sip:192.168.50.60;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" sip:1002@192.168.50.65;tag=tZQ393yprj7BK To: sip:1001@192.0.2.110;transport=ws Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: sip:mod_sofia@192.168.50.65:15060 User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~ b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: sip:1002@192.168.50.65;party=calling;screen=yes; privacy=off
Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_ALERT", "Its a Branch, go manage that !!!<$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY");
} } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!!<$ru>\n"); t_on_failure("MANAGE_FAILURE"); } }
$fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit;}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT:
<script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ? _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com