[svn-commits] mjordan: branch 1.8 r417587 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: mjordan
Date: Sun Jun 29 22:20:12 2014
New Revision: 417587

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=417587
Log:
chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line

This patch is essentially a backport of a small portion of r397526 from
ASTERISK-21981. In that patch, pass through support and format attribute
negotiation was added for Opus. Part of that included being more tolerant to
whitespace in the fmtp line of an SDP; that part of the patch is being
applied here.

As the author of the backport pointed out, in SDP, the fmtp line is allowed to
include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001)
includes an example for this. This was not removed in the updated RFC 4867 in
2007.

Note that this patch only applies to audio in Asterisk 1.8, which is a bit more
limited in its support for format attributes. It does have limited support for
some codecs, so this patch is still useful in this version.

Review: https://reviewboard.asterisk.org/r/3658

ASTERISK-23916
Reported by: Alexander Traud
patches:
  sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520)


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=417587&r1=417586&r2=417587
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Sun Jun 29 22:20:12 2014
@@ -10110,7 +10110,7 @@
 			if (debug)
 				ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
 		}
-	} else if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
+	} else if (sscanf(a, "fmtp: %30u %63[^\t\n]", &codec, fmtp_string) == 2) {
 		struct ast_rtp_payload_type payload;
 
 		payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec);