[asterisk-commits] mjordan: trunk r418868 - in /trunk: UPGRADE.txt channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Jul 17 16:04:07 CDT 2014


Author: mjordan
Date: Thu Jul 17 16:04:01 2014
New Revision: 418868

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=418868
Log:
chan_sip: Make progressinband=never really mean 'never'

progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.

This patch changes the behaviour of this setting in the following ways:

1) In sip_write(), do not pass the media unless we have either progressed
   beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
   media is both set-up and wanted. This helps resolve double-ringing on some
   buggy handsets.

2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
   SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
   enabling early media. Avoid sending double ring indications.

NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.

Review: https://reviewboard.asterisk.org/r/3700

ASTERISK-23972 #close
Reported by: Steve Davies
patches:
  inband_never_present_early_media2 uploaded by Steve Davies (License 5012)


Modified:
    trunk/UPGRADE.txt
    trunk/channels/chan_sip.c

Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=418868&r1=418867&r2=418868
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Thu Jul 17 16:04:01 2014
@@ -198,6 +198,13 @@
    hash to be specified for the DTLS fingerprint placed in SDP. Supported
    values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
 
+ - The 'progressinband=never' option is now more zealous in the persecution of
+   progress messages coming from Asterisk. Channels bridged with a SIP channel
+   that has 'progressinband=never' set will not be able to forward their
+   progress indications through to the SIP device. chan_sip will now turn such
+   progress indications into a 180 Ringing (if a 180 has not yet been
+   transmitted) if 'progressinband=never'.
+
 CLI commands:
  - "core show settings" now lists the current console verbosity in addition
    to the root console verbosity.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=418868&r1=418867&r2=418868
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Jul 17 16:04:01 2014
@@ -7428,8 +7428,11 @@
 						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 					}
 				}
-				p->lastrtptx = time(NULL);
-				res = ast_rtp_instance_write(p->rtp, frame);
+				if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
+									 ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
+					p->lastrtptx = time(NULL);
+					res = ast_rtp_instance_write(p->rtp, frame);
+				}
 			}
 			sip_pvt_unlock(p);
 		}
@@ -7446,8 +7449,11 @@
 					transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
 					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 				}
-				p->lastrtptx = time(NULL);
-				res = ast_rtp_instance_write(p->vrtp, frame);
+				if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
+									 ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
+					p->lastrtptx = time(NULL);
+					res = ast_rtp_instance_write(p->vrtp, frame);
+				}
 			}
 			sip_pvt_unlock(p);
 		}
@@ -7467,8 +7473,11 @@
 						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
 						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 					}
-					p->lastrtptx = time(NULL);
-					res = ast_rtp_instance_write(p->trtp, frame);
+					if (p->invitestate > INV_EARLY_MEDIA || (p->invitestate == INV_EARLY_MEDIA &&
+										 ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT))) {
+						p->lastrtptx = time(NULL);
+						res = ast_rtp_instance_write(p->trtp, frame);
+					}
 				}
 			}
 			sip_pvt_unlock(p);
@@ -7896,8 +7905,14 @@
 		    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
 		    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 			p->invitestate = INV_EARLY_MEDIA;
-			transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
-			ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+			/* SIP_PROG_INBAND_NEVER means sending 180 ringing in place of a 183 */
+			if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_NEVER) {
+				transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
+				ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+			} else if (ast_channel_state(ast) == AST_STATE_RING && !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;
 		}
 		res = -1;
@@ -23002,6 +23017,8 @@
 			if (!req->ignore && p->owner) {
 				/* Queue a progress frame only if we have SDP in 180 or 182 */
 				ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+				/* We have not sent progress, but we have been sent progress so enable early media */
+				ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 			}
 			ast_rtp_instance_activate(p->rtp);
 		}
@@ -23085,6 +23102,8 @@
 			if (!req->ignore && p->owner) {
 				/* Queue a progress frame */
 				ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+				/* We have not sent progress, but we have been sent progress so enable early media */
+				ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 			}
 			ast_rtp_instance_activate(p->rtp);
 		} else {




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