[asterisk-commits] oej: branch oej/roibos-cng-support-1.8 r376466 - in /team/oej/roibos-cng-supp...

[SVN commits to the Asterisk project]

Author: oej
Date: Mon Nov 19 04:04:41 2012
New Revision: 376466

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=376466
Log:
Complete SDP negotiation for both incoming and outbound calls. 

It sounds very funny to have musiconhold playing when I'm not talking. 

Modified:
    team/oej/roibos-cng-support-1.8/channels/chan_sip.c
    team/oej/roibos-cng-support-1.8/main/rtp_engine.c

Modified: team/oej/roibos-cng-support-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/roibos-cng-support-1.8/channels/chan_sip.c?view=diff&rev=376466&r1=376465&r2=376466
==============================================================================
--- team/oej/roibos-cng-support-1.8/channels/chan_sip.c (original)
+++ team/oej/roibos-cng-support-1.8/channels/chan_sip.c Mon Nov 19 04:04:41 2012
@@ -5703,6 +5703,10 @@
 		dialog->noncodeccapability |= AST_RTP_DTMF;
 	else
 		dialog->noncodeccapability &= ~AST_RTP_DTMF;
+
+	if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_ALLOW_CN)) {
+		dialog->noncodeccapability |= AST_RTP_CN;
+	}
 	dialog->directmediaha = ast_duplicate_ha_list(peer->directmediaha);
 	if (peer->call_limit)
 		ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
@@ -9609,7 +9613,7 @@
 		   they are acceptable */
 		p->jointcapability = newjointcapability;                /* Our joint codec profile for this call */
 		p->peercapability = newpeercapability;                  /* The other side's capability in latest offer */
-		p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */
+		p->jointnoncodeccapability = newnoncodeccapability;     /* CN and DTMF capabilities */
 
 		/* respond with single most preferred joint codec, limiting the other side's choice */
 		if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
@@ -11856,10 +11860,12 @@
 				add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
 		}
 
-		/* Now add DTMF RFC2833 telephony-event as a codec */
+		/* Now add Comfort Noise and DTMF RFC2833 telephony-event as a codec */
 		for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
-			if (!(p->jointnoncodeccapability & x))
+			if (!(p->jointnoncodeccapability & x)) {
+				ast_debug(1, "NOT Adding non-codec 0x%x (%s) to SDP\n", x, ast_rtp_lookup_mime_subtype2(0, x, 0));
 				continue;
+			}
 
 			add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
 		}
@@ -19061,6 +19067,7 @@
 			ast_cli(a->fd, "  Format:                 %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
 			ast_cli(a->fd, "  T.38 support            %s\n", AST_CLI_YESNO(cur->udptl != NULL));
 			ast_cli(a->fd, "  Video support           %s\n", AST_CLI_YESNO(cur->vrtp != NULL));
+			ast_cli(a->fd, "  Comfort Noise support   %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[1],SIP_PAGE2_ALLOW_CN)));
 			ast_cli(a->fd, "  MaxCallBR:              %d kbps\n", cur->maxcallbitrate);
 			ast_cli(a->fd, "  Theoretical Address:    %s\n", ast_sockaddr_stringify(&cur->sa));
 			ast_cli(a->fd, "  Received Address:       %s\n", ast_sockaddr_stringify(&cur->recv));

Modified: team/oej/roibos-cng-support-1.8/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/roibos-cng-support-1.8/main/rtp_engine.c?view=diff&rev=376466&r1=376465&r2=376466
==============================================================================
--- team/oej/roibos-cng-support-1.8/main/rtp_engine.c (original)
+++ team/oej/roibos-cng-support-1.8/main/rtp_engine.c Mon Nov 19 04:04:41 2012
@@ -158,7 +158,7 @@
 	[9] = {1, AST_FORMAT_G722},
 	[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
 	[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
-	[13] = {1, AST_FORMAT_CN},
+	[13] = {0, AST_RTP_CN},
 	[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
 	[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
 	[18] = {1, AST_FORMAT_G729A},