]> git.ipfire.org Git - thirdparty/asterisk.git/commit
res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. 23/3223/1
authorAlexander Traud <pabstraud@compuserve.com>
Mon, 18 Jul 2016 10:13:25 +0000 (12:13 +0200)
committerAlexander Traud <pabstraud@compuserve.com>
Mon, 18 Jul 2016 10:47:20 +0000 (05:47 -0500)
commite404f51b42d9518dc8f7153678998d81489d4edb
tree451093acdc7aa3a34b570c34d6d895b38adfdc12
parent962c7ef5d91c546794313e01b4c264efdf3d1ead
res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.

With this change, the initial RTP sequence number is randomly chosen not between
0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
counter (ROC) synchronization is not lost for sRTP, when the very first RTP
packets get lost; see http://srtp.sourceforge.net/faq.html#Q6

ASTERISK-26207 #close

Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
res/res_rtp_asterisk.c