[asterisk-commits] twilson: branch 1.8 r293493 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: twilson
Date: Mon Nov  1 09:58:00 2010
New Revision: 293493

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=293493
Log:
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:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=293493&r1=293492&r2=293493
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Mon Nov  1 09:58:00 2010
@@ -10442,6 +10442,7 @@
 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,} };
@@ -10506,6 +10507,7 @@
 	}
 
 	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) {
@@ -10545,12 +10547,27 @@
 		 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)
@@ -10596,13 +10613,6 @@
 		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