[Asterisk-code-review] channels/chan sip: 180 Ringing not sent after 183 Session Pr... (asterisk[13])

Matt Jordan asteriskteam at digium.com
Mon Nov 23 20:54:49 CST 2015


Matt Jordan has posted comments on this change.

Change subject: channels/chan_sip: 180 Ringing not sent after 183 Session Progress
......................................................................


Patch Set 1: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/#/c/1696/1/channels/chan_sip.c
File channels/chan_sip.c:

Line 7653: 				/* Well, if it's not reasonable, just send in-band */
         : 				if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES && !ast_test_flag(&p->flags[0], SIP_RINGING)) {
         : 					transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
         : 					ast_set_flag(&p->flags[0], SIP_RINGING);
         : 					break;
         : 				}
I'm not sure about this. If we've sent a 183 with an SDP, and then send a 180, I am *sure* there will be some SIP UA out there that will positively melt down. That can absolutely happen here even if we've sent PROGRESS before. In particular, this is like to occur with the default value of progressinband being SIP_PROG_INBAND_NO.

What bug is this attempting to fix? In what scenario when we've already sent a 183 do you expect us to later send a 180?


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Gerrit-MessageType: comment
Gerrit-Change-Id: I498fed853128831b80536f8818cd7b60e641f39c
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Morten Tryfoss <morten at tryfoss.no>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-HasComments: Yes



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