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authorDavid Vossel <dvossel@digium.com>2011-04-18 13:42:51 +0000
committerDavid Vossel <dvossel@digium.com>2011-04-18 13:42:51 +0000
commit4b4549106bd1c6d699468c5d9863ffde7ba590d1 (patch)
tree2922b8393827d1cd34fd1ddb703fb8d26cb49c50 /main
parent6b01eb3324f8ffcaba3c9c4cc30c8c5de60bd413 (diff)
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
Diffstat (limited to 'main')
-rw-r--r--main/rtp_engine.c4
1 files changed, 2 insertions, 2 deletions
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;