[asterisk-commits] file: branch file/rtp_engine r129727 - /team/file/rtp_engine/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Thu Jul 10 15:08:03 2008
New Revision: 129727

URL: http://svn.digium.com/view/asterisk?view=rev&rev=129727
Log:
Bring back the chan_sip RED part.

Modified:
    team/file/rtp_engine/channels/chan_sip.c

Modified: team/file/rtp_engine/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/file/rtp_engine/channels/chan_sip.c?view=diff&rev=129727&r1=129726&r2=129727
==============================================================================
--- team/file/rtp_engine/channels/chan_sip.c (original)
+++ team/file/rtp_engine/channels/chan_sip.c Thu Jul 10 15:08:03 2008
@@ -5107,7 +5107,7 @@
 		if (p) {
 			sip_pvt_lock(p);
 			if (p->red) {
-//				red_buffer_t140(p->trtp, frame);
+				ast_rtp_red_buffer(p->trtp, frame);
 			} else {
 				if (p->trtp) {
 					/* Activate text early media */
@@ -7092,6 +7092,13 @@
 	p->jointcapability = newjointcapability;	        /* Our joint codec profile for this call */
 	p->peercapability = newpeercapability;		        /* The other sides capability in latest offer */
 	p->jointnoncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
+
+        if (p->jointcapability & AST_FORMAT_T140RED) {
+                p->red = 1;
+                ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
+        } else {
+                p->red = 0;
+        }
 
 	ast_rtp_codecs_payloads_copy(&newaudiortp, &p->rtp->codecs, p->rtp);