Merged revisions 314017 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

3 files changed, 7 insertions, 6 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b1f32c06b..da7a12b4a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9092,8 +9092,8 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
 	} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
 		/* We have a rtpmap to handle */
 		if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
-			if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
-			    ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -1) {
+			if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
+			    ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate))) {
 				if (debug)
 					ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
 				//found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -9179,7 +9179,7 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
 		if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
 			/* Note: should really look at the '#chans' params too */
 			if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
-				if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
+				if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
 					if (debug)
 						ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
 					//found_rtpmap_codecs[last_rtpmap_codec] = codec;
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 4c5753e84..f8bf74931 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -968,7 +968,8 @@ void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct as
  * \param options Optional options that may change the behavior of this specific payload
  *
  * \retval 0 success
- * \retval -1 failure
+ * \retval -1 failure, invalid payload numbe
+ * \retval -2 failure, unknown mimetype
  *
  * Example usage:
  *
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index b79ef0bbb..62a515b24 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -501,8 +501,8 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 		}
 
 		/* if both sample rates have been supplied, and they don't match,
-		                      then this not a match; if one has not been supplied, then the
-				      rates are not compared */
+		 * then this not a match; if one has not been supplied, then the
+		 * rates are not compared */
 		if (sample_rate && t->sample_rate &&
 		    (sample_rate != t->sample_rate)) {
 			continue;