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| author | George Joseph <gjoseph@digium.com> | 2017-06-14 16:05:58 -0500 |
|---|---|---|
| committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-06-14 16:05:58 -0500 |
| commit | 8a858275eb55babf03b5b9c6129deee399ce3583 (patch) | |
| tree | 58e0e0e491b0c8ed0339a9937052a62dd70b0cc3 | |
| parent | ffedf681362a55fa8f7339dcbd07dd576fdc7877 (diff) | |
| parent | 7dafe82751fd512d58bb3843601daff013958dd2 (diff) | |
Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp" into 13
| -rw-r--r-- | res/res_rtp_asterisk.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index f75482f7f..5003c95d7 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -3443,7 +3443,7 @@ static void ast_rtp_change_source(struct ast_rtp_instance *instance) 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(srtp, rtp->ssrc, ssrc); + res_srtp->change_source(rtcp_srtp, rtp->ssrc, ssrc); } } |