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Merged revisions 284477 via svnmerge from

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

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  r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
  
  Fix SRTP for changing SSRC and multiple a=crypto SDP lines
  
  Adding code to Asterisk that changed the SSRC during bridges and masquerades
  broke SRTP functionality. Also broken was handling the situation where an
  incoming INVITE had more than one crypto offer. This patch caches the SRTP
  policies the we use so that we can change the ssrc and inform libsrtp of the
  new streams. It also uses the first acceptable a=crypto line from the incoming
  INVITE.
  
  (closes issue #17563)
  Reported by: Alexcr
  Patches: 
        srtp.diff uploaded by twilson (license 396)
  Tested by: twilson
  
  Review: https://reviewboard.asterisk.org/r/878/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-09-01 18:52:27 +00:00
parent af67f7621a
commit 920f5ea8b7
5 changed files with 110 additions and 9 deletions
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12
res/res_rtp_asterisk.c
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@@ -717,12 +717,24 @@ static void ast_rtp_update_source(struct ast_rtp_instance *instance)
static void ast_rtp_change_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
unsigned int ssrc = ast_random();
if (!rtp->lastts) {
ast_debug(3, "Not changing SSRC since we haven't sent any RTP yet\n");
return;
}
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
if (srtp) {
ast_debug(3, "Changing ssrc for SRTP from %u to %u\n", rtp->ssrc, ssrc);
res_srtp->change_source(srtp, rtp->ssrc, ssrc);
}
rtp->ssrc = ssrc;
return;