From: Terry Wilson <twilson@digium.com>
Date: Mon, 17 Jan 2011 17:45:39 +0000 (+0000)
Subject: Merged revisions 293493 via svnmerge from
X-Git-Tag: 1.4.41-rc1~44
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=59bd4a2ba3a2f18bb105635de27167a977849609;p=thirdparty%2Fasterisk.git

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 0017403)
  Reported by: one47
  Patches:
        sip_codecs_simplified4 uploaded by one47 (license 23)
  Tested by: one47, falves11

  Review: https://reviewboard.asterisk.org/r/967/ [^]
........

Back port a fix that should have been included


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@302087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 993ae5ae9b..4ec0a51765 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7027,6 +7027,7 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int add_audio, int add_t38)
 {
 	int alreadysent = 0;
+	int doing_directmedia = FALSE;
 
 	struct sockaddr_in sin;
 	struct sockaddr_in vsin;
@@ -7092,6 +7093,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 	if (p->redirip.sin_addr.s_addr) {
 		dest.sin_port = p->redirip.sin_port;
 		dest.sin_addr = p->redirip.sin_addr;
+		doing_directmedia = p->redircodecs ? TRUE : FALSE;
 	} else {
 		dest.sin_addr = p->ourip;
 		dest.sin_port = sin.sin_port;
@@ -7108,15 +7110,21 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		hold = "a=sendrecv\r\n";
 
 	if (add_audio) {
+		char codecbuf[SIPBUFSIZE];
 		capability = p->jointcapability;
 
-
 		if (option_debug > 1) {
-			char codecbuf[SIPBUFSIZE];
 			ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
 			ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
 		}
 
+		if (doing_directmedia) {
+			capability &= p->redircodecs;
+			if (option_debug > 1) {
+				ast_log(LOG_NOTICE, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability));
+			}
+		}
+
 #ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
 		if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) {
 			ast_build_string(&m_audio_next, &m_audio_left, " %d", 191);