[svn-commits] twilson: trunk r302048 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: twilson
Date: Mon Jan 17 10:38:21 2011
New Revision: 302048

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=302048
Log:
Merged revisions 293493 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines
  
  Only offer codecs both sides support for directmedia
  
  When using directmedia, Asterisk needs to limit the codecs offered to just
  the ones that both sides recognize, otherwise they may end up sending audio
  that the other side doesn't understand.
  
  (closes issue #17403)
  Reported by: one47
  Patches: 
        sip_codecs_simplified4 uploaded by one47 (license 23)
  Tested by: one47, falves11
  
  Review: https://reviewboard.asterisk.org/r/967/
........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=302048&r1=302047&r2=302048
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jan 17 10:38:21 2011
@@ -10597,6 +10597,7 @@
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
 {
 	format_t alreadysent = 0;
+	int doing_directmedia = FALSE;
 
 	struct ast_sockaddr addr = { {0,} };
 	struct ast_sockaddr vaddr = { {0,} };
@@ -10661,6 +10662,7 @@
 	}
 
 	if (add_audio) {
+		doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && p->redircodecs) ? TRUE : FALSE;
 		/* Check if we need video in this call */
 		if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
 			if (p->vrtp) {
@@ -10700,12 +10702,27 @@
 		 ast_sockaddr_stringify_addr(&dest));
 
 	if (add_audio) {
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) {
+			hold = "a=recvonly\r\n";
+			doing_directmedia = FALSE;
+		} else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE) {
+			hold = "a=inactive\r\n";
+			doing_directmedia = FALSE;
+		} else {
+			hold = "a=sendrecv\r\n";
+		}
+
 		capability = p->jointcapability;
 
 		/* XXX note, Video and Text are negated - 'true' means 'no' */
 		ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
 			  p->novideo ? "True" : "False", p->notext ? "True" : "False");
 		ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
+
+		if (doing_directmedia) {
+			capability &= p->redircodecs;
+			ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability));
+		}
 
 		/* Check if we need audio */
 		if (capability & AST_FORMAT_AUDIO_MASK)
@@ -10751,13 +10768,6 @@
 		get_crypto_attrib(p->srtp, &a_crypto);
 		ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ast_sockaddr_port(&dest),
 			a_crypto ? "SAVP" : "AVP");
-
-		if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
-			hold = "a=recvonly\r\n";
-		else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
-			hold = "a=inactive\r\n";
-		else
-			hold = "a=sendrecv\r\n";
 
 		/* Now, start adding audio codecs. These are added in this order:
 		   - First what was requested by the calling channel