[Asterisk-code-review] res_rtp_asterisk: Always return provided DTLS packet length. (asterisk[13])

[Joshua Colp]

Joshua Colp has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/13177 )


Change subject: res_rtp_asterisk: Always return provided DTLS packet length.
......................................................................

res_rtp_asterisk: Always return provided DTLS packet length.

OpenSSL can not tolerate if the packet sent out does not
match the length that it provided to the sender. This change
lies and says that each time the full packet was sent. If
a problem does occur then a retransmission will occur as
appropriate.

ASTERISK-28576

Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449
---
M res/res_rtp_asterisk.c
1 file changed, 8 insertions(+), 2 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/77/13177/1

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index cb68abe..f2601ed 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -589,6 +589,10 @@
 	struct ast_sockaddr remote_address = { {0, } };
 	int ice;
 
+	/* OpenSSL can't tolerate a packet not being sent, so we always state that
+	 * we sent the packet. If it isn't then retransmission will occur.
+	 */
+
 	if (rtp->rtcp && rtp->rtcp->dtls.write_bio == bio) {
 		rtcp = 1;
 		ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
@@ -597,10 +601,12 @@
 	}
 
 	if (ast_sockaddr_isnull(&remote_address)) {
-		return 0;
+		return len;
 	}
 
-	return __rtp_sendto(instance, (char *)buf, len, 0, &remote_address, rtcp, &ice, 0);
+	__rtp_sendto(instance, (char *)buf, len, 0, &remote_address, rtcp, &ice, 0);
+
+	return len;
 }
 
 static long dtls_bio_ctrl(BIO *bio, int cmd, long arg1, void *arg2)

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449
Gerrit-Change-Number: 13177
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-MessageType: newchange
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