diff options
| author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-01 18:52:27 +0000 |
|---|---|---|
| committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-01 18:52:27 +0000 |
| commit | 80c977d645454f2d49046399944d014b4071a179 (patch) | |
| tree | d44985b2775c7b67a6c6b95e497c7f9941721451 /res/res_rtp_asterisk.c | |
| parent | 6999ba0d4ad85f6434e765d858aec69b7caf13cd (diff) | |
Merged revisions 284477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
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/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284479 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res/res_rtp_asterisk.c')
| -rw-r--r-- | res/res_rtp_asterisk.c | 12 |
1 files changed, 12 insertions, 0 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index df7cf27b5..6ddd33902 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -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; |