Summary:	ASTERISK-06498: Asterisk is not constructing the BYE request RURI correctly from contact headers in some situations.
Reporter:	aubergine (aubergine)	Labels:
Date Opened:	2006-03-07 18:23:31.000-0600	Date Closed:	2006-04-07 03:20:37
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) bug_6674.txt
Description:	For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs it from scratch. This is of particular concern when one is using a SIP clients that only do TCP like office communicator (and of course a proxy in the middle to do UDP/TCP protocol conversion) . In these cases well-behaving UA puts a transport=tcp tag into the contact header of either the INVITE or the 200 OK. In the first scenario this tag is preserved and the BYE's RURI has the transport tag (which the proxy can use for protocol conversion). The second scenario loses the transport tag in the BYE's RURI (and the proxy has no idea that protocol conversion is needed) ****** ADDITIONAL INFORMATION ****** parse_ok_contact() is only called from handle_response_invite() which is called when * gets a 200 OK from an INVITE. This handle the above first scenario correctly I naively add a parse_ok_contact() to handle_request() ie:        case SIP_INVITE:                parse_ok_contact(p, req);                res = handle_request_invite(p, req, debug, ignore, seqno, sin, recount, e);                break; and this seems to handles the second scenario correctly. However parse_ok_contact() does a lot more than populate p->okcontacturi
Comments:	By: Olle Johansson (oej) 2006-03-08 07:02:02.000-0600 Please try this patch. I haven't tested it, so I am grateful for any tests, debug logs etc. By: mikma (mikma) 2006-03-10 12:06:59.000-0600 I'm also being hit by this bug when I call Asterisk via Openser and the calling UA is connected to Openser with TLS and has included ";transport=TLS" in the Contact uri. The following small patch seems to solve the problem. What's the reason to remove the uri parameters from the Contact uri? --- channels/chan_sip.c (revision 12340) +++ channels/chan_sip.c (arbetskopia) @@ -5007,9 +5030,9 @@        char *c, *n;        ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));        c = get_in_brackets(stripped); -       n = strchr(c, ';'); -       if (n) -               *n = '\0'; +/*     n = strchr(c, ';'); */ +/*     if (n) */ +/*             *n = '\0'; */        if (!ast_strlen_zero(c))                ast_string_field_set(p, uri, c); } By: aubergine (aubergine) 2006-03-13 18:17:28.000-0600 What version is this patch against? By: aubergine (aubergine) 2006-03-14 01:33:39.000-0600 I've patched 1.2.3 and it appears to be working. Also FYI set_adddress_from_contact() has too many 'd's. Thanks for the patch By: Olle Johansson (oej) 2006-04-07 03:20:07 Committed to svn trunk. Thanks for testing.