diff options
| author | Joshua Colp <jcolp@digium.com> | 2018-04-30 14:38:36 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2018-05-01 10:52:34 -0600 |
| commit | afdca5c68c6354ada43f341666d0cc9e14bf9fee (patch) | |
| tree | 0505e52e577879a9ccc066933b2563a6bf4f0a0e /res/res_rtp_asterisk.c | |
| parent | f633af89c1d69734589dac2cb16f6de46301c9d6 (diff) | |
res_rtp_asterisk: Always update SRTP on local SSRC change.
When the local SSRC changes we need to update the SRTP information
so that the proper key is used. This is commonly done as a result
of bridging two channels together. Previously we only updated
the SRTP information if media had already flowed, but in practice
the channel driver may have already performed SRTP negotiation and
set up the previous SSRC. We now always do it on a local SSRC
change.
ASTERISK-27795
ASTERISK-27800
Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10
Diffstat (limited to 'res/res_rtp_asterisk.c')
| -rw-r--r-- | res/res_rtp_asterisk.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index 4ac20d551..5579914af 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -3947,15 +3947,15 @@ static void ast_rtp_change_source(struct ast_rtp_instance *instance) if (rtp->lastts) { /* 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); + 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); - if (rtcp_srtp != srtp) { - res_srtp->change_source(rtcp_srtp, 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); + if (rtcp_srtp != srtp) { + res_srtp->change_source(rtcp_srtp, rtp->ssrc, ssrc); } } |